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OpenStage 5 SIP OpenScape Voice

1. sessssssss 50 Date and time tet ac dee ended 40 E Download application sseesessesssssss 46 Environment depending uicina 133 Emergency number sssss 95 LAN erc seas 37 Error Messages nM AE AAAA E E E E DDR 130 Miscellaneous sees 40 Errors Network addresses iuis etes ikea aenea eda 35 Dne featecctace detent d map tute tiui te uM gs 129 Quality Of SERVICE M 36 Explanations to specialized terms 125 Restore factory settings ssssessss 53 pm 50 SIP parameter siinon priusa 38 F SNMP iiiter a e eai aa de rr ies 49 Software update sesssssseseen 44 Factory settings restore cccccceeeeeeeeeeeeeseneeees 53 Speech parameter sss 50 Fault Finding ccccececeeeeeeeeeeeeeneeceeeeeeeeeeteeeteees 129 Status of transferred files sssessssse 48 Feature toggle e eir ni e rene ttes 96 Terminal details esses 37 FTP Configuration download filename 88 ACCOUNT NAME tierra ai 96 CC OIMORMMNLY iiic deg essan idc dit rb A 5 cU 97 Control K yS ooi et eret et bei ein 15 xu M 97 Server requirements sssssssssssss 45 BETIS 97 D Function Key ssssssseeene eene 97 Function Keys iie o
2. ssssssssseseee 114 M SIP MAD address iei caulis 105 Addresses 4 ettet ende dete e iens 114 Maintenance of the phone es 4 Ae amt 115 Message Waiting IP address 105 Auto reconnect ssssssssssseeeeneee 115 Beep on auto answer sssssssee 115 Beep on auto reconnect cccceeeeeeeeeteeeeees 115 ieu 19 GNI P 18 Index 139 PasswWoI eo ar Exe p ER De 115 Time configuration sesse aeae 40 clum 115 Time zone offset 1 dete edd uas 122 ROWING TP 116 HIN 129 Server tyDOe ideo ab e pd eK rp aad 116 Transferring files eeeeeeseeseeeee 44 Sennels cause br dence Ep a 19 Troubleshooting seeene 129 Session timer enabled sseusssssss 117 Session timer value sssssssssssssss 116 Transport protocol sssssssssssss 117 U User ID et eR e ie aiaa 117 Ec T sii Enta 117 Update software sss 44 SNMP Upload configuration ssssesssssss 123 DNS nale sion tese ie Caco E Non es 118 Upload download status ssssse 123 Overview E 25 Used symbols ssseeeeeeeenennn 8 Pass VO f j ec s 118 User SUpport nadora hinaia aaia aa aai 9 Trap IP address sess 118 SNTP DNS NAME coe coche sad ud ce LE 18 V P address EPUM d ME CN D TIS Versions Info 2 e HE 123
3. 2 Enter the administrator password default 123456 max length 24 digits and confirm Administrations Menu e General Information Network IP and Routing System o SIP environment o SIP features Quality of service File transfer and phone download settings Time and date SNMP Speech Ringer settings LAN Port settings Multiline operation Function keys o Phone Dial plan Feature Access e Configuration Management o Settings o Check for updates o Error log Upload Download o Upload configuration o Download application o Download configuration o Download hold music e Diagnostics and statistics Non user assisted tests User assisted tests RTP Statistics QoS Data Collection Fault investigation ooog o Simplified trace page Security Restart terminal Reset user password Change admin password Clear all user data Restore factory setting Port Control FPN Port Settings Survivability Home Web Interface 61 Web Pages D Click on the required field to navigate to the description of a parameter e g move cursor over Application 2 3 7 and press the left mouse button to get to the descripton in the alpha betical reference chapter The links after the symbol A lead to the administration tasks with menu paths If DNS is applicable Page 24 the fields for entering the IP addresses on the following web pag es have the addition or DNS name gt SI
4. IP Abbreviation for Internet Protokoll IP address Also called IP in short The unique address of a terminal device in the network It consists of four number blocks of 0 to 255 each separated by a point To simplify the notation voice names can be released from a gt DNS into the IP addresses Jitter Runtime fluctuations in data transmission in IP networks LAN Abbreviation for Local Area Network Layer 2 2nd layer Data Link Layer of the 7 layer OSI model for describing data transmission interfaces Layer 3 3rd layer Network Layer of the 7 layer OSI model for describing the data transmission interfaces LDAP Abbreviation for Lightweight Directory Access Protocol Simplified protocol for accessing standardized directory systems e g a company telephone di rectory LED Abbreviation for Light Emitting Diode Cold light illumination in different colours at low power consumption MAC Abbreviation for Medium Access Control Address A 48 bit address with the help of which a terminal device e g gt IP telephone or Network card identifies itself uniquely in a network all over the world MIB Abbreviation for Management Information Base Database containing descriptions of error messages of the devices and functions in a network Alphabetical Reference 127 PBX Abbreviation for Private Branch eXchange Private telephone system that connects the different internal devices to t
5. Other settings All other settings of your OpenStage 5 SIP must be made through the Web based Management Tool Page 59 Example 1 enter the IP address 192 168 1 44 To enter the ASCII code of the IP address follow the steps below Press the keys successively all LEDs flash Enter admin password default 123456 Terminate the operation You are now in the Administration Area Press 1 function key to make settings Enter code for IP address Enter character 1 Enter character 9 Enter character 2 Enter character dot Enter character 1 Enter character 8 Enter character 6 Enter character dot Enter character 1 32 Basic Administration 26006 e vw M oo Enter character dot Enter character 4 Enter character 4 Terminate the operation If this is the last operation at all don t forget to confirm your entries and restart the telefon see page before Example 2 Check the IP address If you want to do other settings through the Web based Management Tool you have to know the current IP address To find out the address in binary code fol low the steps below Press the keys successively all LEDs flash Enter admin password default 123456 Terminate the operation You are now in the Administration Area Press 2 function key to view settings results ASCII codes see gt Page 81 Enter code for IP address Step to the first character Upper Byte shows 3
6. Reorient or relocate the receiving antenna Increase the separation between the equipment and receiver Connect the equipment into an outlet on a circuit different from that to which the receiver is connected Consult the dealer or an experienced radio TV technician for help This product is a UL Listed Accessory I T E in U S A and Canada Location of the Telephone The telephone should be operated in a controlled environment with an ambient temperature be tween 5 C and 40 C 41 F and 104 F To ensure good handsfree talking quality the area in front of the microphone front right should be kept clear The optimum handsfree distance is 20 inches 50cm Do not install the telephone in a room where large quantities of dust accumulate this can con siderably reduce the service life of the telephone Do not expose the telephone to direct sunlight or any other source of heat as this is liable to damage the electronic equipment and the plastic casing Do not operate the telephone in damp environments such as bathrooms Telephone Maintenance Always use a damp or antistatic cloth to clean the telephone Never use a dry cloth f the telephone is very dirty clean it with a diluted neutral cleaner containing some form of sur factant such as a dish detergent Afterwards remove all traces of the cleaner with a damp cloth using water only Never use cleaners containing alcohol cleaners that corrode plastic
7. IP address Gateway Mask W gt Page 99 sere IP and routing IP routing Route 1 2 Route enter IP address Gateway enter IP address Mask enter subnet mask DNS Domain Name XW gt Page 94 e Network IP and routing Domain name enter domain name Primary DNS IP Address W gt Page 108 e Network IP and routing Primary DNS IP address enter IP address Secondary DNS IP Address W gt Page 113 e Network IP and routing Secondary DNS IP address enter IP address Terminal hostname W gt Page 120 e Network IP and routing Terminal hostname change name Use dynamic hostname W gt Page 123 e Network IP and routing Use dynamic hostname concept mark to enable disable NAT keep alive W gt Page 105 e Network IP and routing NAT keep alive range 10 to 3600 D Changing either the DHCP IP assignment or the Terminal IP address will take effect as soon as the OpenStage 5 SIP is restarted 36 Extended Administration Quality of Service QoS By changing the Quality of Service parameter you can affect the speech quality results Further speech quality parameters see Page 50 QoS Configuration Parameter QoS Mode W gt Page 109 Web Interface path Menu gt Page 60 e Quality of Service Required select QoS mode Layer 3 Voice only if L3On W gt Page 99 e Quality of Service Layer 3 Voice se
8. SIP features SIP Features Page 39 Transfer on Ringing Deactivated 20 Quality of Service Quality of Service QoS gt Page 36 Quality of Service QoS WARNING If you make changes to any of the fields on this page you will have to restart the terminal manually before they take effect Web Interface 65 File transfer Software Update Transferring Files gt Page 44 WARNING Note that file transfers to and from the download server will take place over an insecure link 0 0 0 0 guest guest OS5a app OS5n i lOSSc jOS5 moh No transfer Time and date Al Configuring Date and Time gt Page 40 a Jaway v DD MM YY v 66 Web Interface SNMP Use SNMP gt Page 49 SNMP Settings Speech Al Change Speech Parameters gt Page 50 6 729 x GrMPrefemed v 20ms a Web Interface 67 Ringer settings Configure Ringer Settings gt Page 50 Ringer Settings lert external elcore dr2 lert internal JANI ert priority LAN port settings Al LAN Port Settings gt Page 37 LAN Port Settings Web Interface Multiline operation A Multiline gt Page 41 Multiline Operation Originating line preference idle line gt Terminating line preference Ringing line z Line key operation made Hoa z Rollover type Alert beep z Rollover volume 2zl Registration LEDs Iv Show f
9. USA S30122 H7724 X IM S30122 H7726 X See also http wiki unify com wiki Power supply and PoE classes Plug the plug in power supply unit into the mains Plug the connector 3 at the bottom of the telephone into the plug in power supply unit Plug the jack of the LAN cable into the connector 1 at the bottom of the telephone and connect the cable with LAN Feed the cables through the relief on the back of the housing and fix them by means of the cable clip Power over LAN information Power over LAN support is provided on the LAN port and complies with the IEEE802 3af standard 8 wire Ethernet cables are required to use it 12 Startup Procedure Start Power on Reboot Key 3 pressed o Y Application is starting Ye M Netboot request see http wiki unify com No Yes Netboot Upgrade Run up and wait 120s DHCP No activated A DHCP Discover in untagged LAN T Y L No lt gt Yes VLAN ID in Option Yes Y DHCP Discover in VLAN Yes Registration No L2 activated v Using manual configuration v DHCP Discover bh in untagged LAN DHCP Discover in tagged LAN Yes Run up and wait 120s Installation 13 Usi
10. If DHCP is enabled this field is provided automaticly It is not writeable Al gt Page 35 1 gt Page 62 Download Application Use this function to download an updated software version for the OpenStage 5 SIP from the gt FTP server The following parameters must be set before undertaking the download operation gt Download server IP address or DNS name gt FTP path gt Application filename gt FTP account name gt FTP username gt FTP password Detailed description gt Page 44 Al gt Page 46 fF gt Page 65 Download Configuration Use this function to download a configuration for the OpenStage 5 SIP stored on the gt FTP server The following parameters must be set before undertaking the download operation gt Download server IP address or DNS name gt FTP path gt Configuration filename gt FTP account name gt FTP username gt FTP password A Page 46 3 gt Page 65 Download Netboot Use this function to download the netboot file for the OpenStage 5 SIP stored on the gt FTP server The following parameters must be set before undertaking the download operation gt Download server IP address or DNS name gt FTP path gt Netboot filename gt FTP account name gt FTP username gt FTP password Al gt Page 46 F3 gt Page 65 Download server IP address or DNS name Enter the gt IP address or host name of the gt FTP server to upload and
11. Multiline operation Show focus mark to enable Forwarding Indication 27 gt Page 96 e Multiline operation Use LED to indicate Remote Forwarding mark to enable Reservation Timer 27 gt Page 112 US Multiline operation Reservation timer set time Dial Plan Configuration and Status Parameter Dial Plan W gt Page 90 Web Interface path Menu gt Page 60 e System Dial Plan Action enable disable Dial Plan Info W gt Page 90 l General information Dial plan Name and Status Extended Administration 43 Direct Station Select DSS Each DSS key will be a special variant of a line key The configuration specifies whether a line key will be a DSS key or a normal multiline key The system operation and protocol of the DSS key will be the same as for a line key and the OpenScape Voice will not be required to know if a line ap pearance is associated with a DSS key or a multiline key A DSS key will use the line key mechanism to display the line state via the LED associated with the key However the DSS key will only present a subset of the line states to the user i e Idle Alerting and Busy All other states that a Keyset line key could present will be forced into one of the valid DSS states A major departure from Keyset line key operation is the action taken when a DSS key is pressed The DSS action falls into two basic camps 1 Pickup a call alerting
12. Parameter Terminal Number W gt Page 121 Web Interface path Menu Page 60 e SIP environment Terminal number enter terminal number SIP User ID W gt Page 117 SIP environment SIP user ID enter ID SIP Password W gt Page 115 e SIP environment New Confirm SIP password enter re enter password SIP Server Address W gt Page 114 e SIP environment Server IP address enter IP address SIP Registrar Address W gt Page 114 e SIP environment I Registrar IP address enter IP address SIP Routing W gt Page 116 SIP environment SIP routing select routing Extended Administration 35 Extended Administration Configure Network Parameters To access a SIP server as an IP client some network related information have to be configured Ip Depending on the SIP network environment different changes are necessary Page 29 Network Addresses Parameter DHCP IP Assignment W gt Page 89 Web Interface path Menu gt Page 60 e Network IP and routing DHCP activate deactivate checkbox Terminal IP Address W gt Page 121 e Network IP and routing Terminal IP address enter IP address Terminal Mask W gt Page 121 e Network IP and routing Terminal mask enter terminal mask Default Route Gateway W gt Page 89 e Network IP and routing Default gateway enter gateway address IP Route 1 2
13. SNMP settings Send QDC Taps to Management Center enable disable Queries Allowed W gt Page 118 1 or host name if DNS is applicable View SNMP Errors Parameter MIB2 Discards W gt Page 93 e SNMP settings Queries Allowed enable disable Page 24 Web Interface path Menu gt Page 60 c SNMP SNMP MIB2 errors MIB2 Err Count W gt Page 99 e SNMP SNMP MIB2 errors 50 Extended Administration Change Speech Parameters Parameter Web Interface path Menu gt Page 60 Audio Mode e W gt Page 84 Speech Audio mode select audio mode Compression Encoding c W gt Page 88 Speech Compression encoding select compression RTP Packet Size e W gt Page 113 Speech RTP packet size select packet size Silence e W gt Page 114 Speech Silence Suppression activate deactivate checkbox Play DTMF RFC 2833 e XW gt Page 108 Speech Play DTMF RFC 2833 activate deactivate checkbox Configure Ringer Settings Audio Visual Indications This setting is used to setup Alert Indications that can be used to differentiate between call types Web Interface path Parameter Menu Page 60 Alert Indications E W gt Page 83 Ringer Settings enter alert indication string and enter mel ody tone and duration IE This feature is only supported in specific system environments Extended Administration
14. Survivability Backup Registration Timer The Backup Reg Timer option displays the duration of the SIP registration requested by the phone when it registers with the backup proxy server Note The phone only registers with the backup proxy if the Backup registration setting is On A gt Page 54 1 gt Page 80 Survivability Backup OBP The Backup OBP flag indicates whether or not the Backup Proxy Server is used as an outbound proxy A gt Page 54 1 gt Page 80 Survivability Backup Transport The Backup transport option displays the current transport protocol used to carry SIP messages to the Backup proxy server Options e TCP UDP UDP is prepared Al gt Page 54 1 gt Page 80 120 Alphabetical Reference Terminal Hostname This field ist provided with the E164 number but you can change it Enter a new hostname for the telephone The hostname is transmitted to the DHCP server together with the MAC address while the telephone registers at the DHCP server The DHCP server sends an IP address to the telephone at the same time it transmits this IP address together with the host name to the DNS server where this association is registered Within the DNS server s range the telephone can now be addressed using its host name If Mobility is unsing the hostname is overwriten by the current E164 number of the Mobility user if the option Use E164 as hostname is enabled The new hostname can be used to open t
15. https address where address is the IP address or host name of the OpenStage 5 SIP You can access the web interface in the browser using the host name assigned to your telephone The presetting for the host name is the current E164 number An example for the browser call is https hostname domainname For configuring the phone s IP address see chapter Basic Administration gt Page 29 For example the configuration page for the Phone with the IP address 192 168 1 137 is https 192 168 1 137 I After entering the URL the browser might display a certificate notification A screen like the following home page appears OpenStage 5 SIP Phone number 3331 Home Page Administration User OpenStage 5 SIP Home Page The OpenStage 5 SIP web pages allow the administrator to configure the administrator settings perform diagnostic tests download new software and the user to assign features to function keys change the sword s hange call related parameters change secure call setting i Click on the required field in the dialogs to see a description for each parameter 60 Web Interface Access to the Web Interface Administrator Menu The following steps describe the access to the administrator menu starting from the home page of the OpenStage 5 SIP 1 Click on the link Administration The following login dialog appears Local administrator login Password
16. latency variations jitter and delay For further information see gt Quality of Service QoS Al gt Page 36 F3 gt Page 64 Layer 3 Voice Can be defined only if the gt Layer 3 support is activated Select the desired value see gt Quality of Service QoS A gt Page 36 3 gt Page 64 100 Alphabetical Reference Layer 3 Signalling Can be defined only if the gt Layer 3 support is activated Select the desired value see gt Quality of Service QoS Al gt Page 36 F3 gt Page 64 Layer 2 Default Can be defined only if the gt Layer 2 support is activated gt Quality of Service QoS Value range table Permitted values numeric Range 0 7 for each 64 positions Default values pos 12 6 pos 18 3 Al gt Page 36 F3 gt Page 64 Layer 2 signalling Can be defined only if the gt Layer 2 support is activated gt Quality of Service QoS Al gt Page 36 F3 gt Page 64 Layer 2 voice Can be defined only if the gt Layer 2 support is activated gt Quality of Service QoS Al gt Page 36 F3 gt Page 64 LAN port settings Use this function to select the LAN port speed Al gt Page 36 F3 gt Page 67 LAN port speed Use this function to define the bandwidth at which the OpenStage 5 SIP should be run The re quired value depends on the bandwidth that the switch or router supports in the network Bandwith Automatic in standard case automa
17. nd ex Asslgnimett 2 2 c dicto tacite bete d 89 Configuration with een 29 Configuration without seeeeeeee 30 A jeu MPEL 22 DHCP IP assignment ssessee 30 Abbreviations iarere cetera a d Rena denne nin 125 DHCP server ooccccccccccccccccccccccccceeceteceeeeeeeeeceeeeseesetseess 22 Action on submit esee 82 Diagnostic tests i e totns an tnus 51 Administrator password sseseeeeee 82 Dial Plan suseteoRtpeo noe en teed 90 Alert indication seeeeeeeeetnnntenntns 83 CENSESINIO agar state necne tun eRu s dbneiga Edda iut 93 Append codes 22 83 Dialling keypad 1 rtt ttn nnn nnns 13 Application Discards in outbound packets u 93 Software download sssssee Ho DES ee ems ium mda 27 Application download filename 84 DNS Application software IP ES 108 113 Update tc ree te lee regedit AERES 44 Overview eee ess 24 Audio loop test ssseen 84 Domain name ucc qeosssceqi ni psestbidoqu Muss SIUE 94 Audio mode uet it ttt dete hee e xa ae 84 Download Conflg tratlon sauce creciente reris 94 IP address DNS name sss 94 C DSS Codec Negotiation Ap MM Compression encoding eene 88 lup 95 Configuration DTMF nren 108 Administrator password sssssesssssss 53 Audio visual indications
18. 2833 LAN Port Settings LAN port 1 LAN Feature access Auto answer CTI on Callback busy on Call join on Call transfer on Do not disturb on Hot keypad dialing on Phone Configurations Parameter Music on hold Callback no replay Call hold explicit GPU New Call Beep Phone function key assignments Message waiting Voice Messages Cancel Release Confirm Blind Transfer Security Settings Payload security allowed Connectivity check interval 0 disabled Port control SIP server validation off Service Agent Test Interface Country Settings SNMP Interface Country United Kingdom 136 Phone Configurations WEB page Parameter Audio Settings Volume Settings Handset Volume Loudspeaker Volume Key Click Volume Rollover Volume Ringer Settings Ringer Volume Melody Tone Sequence Call related parameters Auto dial timer DND feature enabled on Idle dialing mode User Security Setting Audible secure call indicator Information and support for our products can be found on the Internet at http www unify com Technical notes current information about firmware updates frequently asked questions and lots more can be found on the Internet at http wiki unify com Index 137 DHCP
19. General Information HRE t General Information About the Manual The instructions within this manual will help you in administering and maintaining the OpenStage 5 SIP The instructions contain important information for safe and proper operation of the OpenStage 5 SIP Follow them carefully to avoid improper operation and get the most out of your multi function telephone in a network environment This guide is intended for service providers and network administrators who administer VoIP ser vices using the OpenStage 5 SIP and who have a fundamental understanding of SIP The tasks described in this guide are not intended for end users of the phones Many of these tasks affect the ability of a phone to function on the network and require an understanding of IP networking and telephony concepts mp For your own protection please read the section dealing with safety Follow the safety in structions carefully in order to avoid endangering yourself or other persons and to prevent damage to the unit These instructions are laid out in a user oriented manner which means that you are led through the functions of the OpenStage 5 SIP step by step from the setup through descriptions of tools and extensions discussions of special administrative and service tasks at the end of the manual For the users a separate manual is provided Symbols in the Manual Attention This symbol indicates a hazard Failure to follow the i
20. Web Interface path Parameter Menu gt Page 60 Netboot Filename e W gt Page 106 File transfer Netboot filename enter filename Download Netboot Action on submit select Download NETBOOT XW gt Page 94 Extended Administration 47 Port Numbering The phone will provide the ability to configure any TCP or UDP port number that is currently fixed with the exception of SNMP SNTP DNS The table below gives the port addresses that will be configurable fault values see also RFC 1700 Assigned numbers The table also shows the de In some cases a port base number is used The port base number will be the configurable item not the port numbers derived from the port number base Function Default Default Comment Value Value UDP TCP RTP port range local 5010 to 5022 RTP port number is even RTCP port number is RTP port number 1 5004 to 5006 reserved by IANA but can be used by the phone RTP port range remote any RTCP port range local 5011 to 5023 RTP port number is even RTCP port number is RTP port number 1 5005 to 5007 reserved by IANA but can be used by the phone RTCP port range remote any HTTP Hypertext Transfer Protocol 8085 HTTPS Secure Hypertext Transfer 443 Protocol SNMP 161 Not configurable SNMP Traps 162 Not configurable SNTP 123 Not configurable SNTP Heart be
21. a supported function Page 97 A phone where all lines are represented by a line key plus an LED Every keyset will have a primary line and may have sec Neyset ondary or phantom lines 10 Line keys can be configured for a keyset A representation of a valid SIP AoR Address of Record A line Line is the context for connecting SIP calls A line may support one or more calls Line Appearance A line Directory Number that appears on one or more Keyset devices as a primary line currently only one device per primary line secondary Line or phantom line Line key A function key that is used to represent a line appearance or in the future call appearance on a line Consultation hold A form of hold which is private to the holding keyset Manual hold A form of hold which is accessible to any keyset on the same shared line Private line type A line that only is accessed by one SIP endpoint i e it is exclu sively owned Shared line type A line that may be accessed by multiple SIP endpoints INVITEs to a shared line are FORKED to all SIP endpoints sharing the line 102 Alphabetical Reference The line that characterises the oP410 420 phone user Every keyset will have a primary line This line can be expected to use the public DN of the OpenStage 5 SIP phone user There is only one Primary line instance per OpenStage 5 SIP phone A primary line on a different phon
22. by a gt SNTP server Select the date format and enter the date and time information A gt Page 40 3 gt Page 65 Alphabetical Reference 89 Daylight saving If your country uses daylight saving time you have to switch this feature on and off manually twice a year independently whether SNTP is used or not gt Page 118 On means an offset of 1 Off means no offset default A gt Page 40 3 gt Page 65 Default domain name f you use an Outbound Proxy server you can define a valid domain name of this server To use this setting you have to activate the Outbound Proxy option Page 107 Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 38 F3 gt Page 63 Default Route Enter the IP address that was assigned to the router of your IP network if not provided by gt DHCP dynamically gt DHCP IP assignment If the value was assigned dynamically it can only be read The change will only have effect if you restart the phone Value range table Permitted values numeric Length max 15 digits incl dots Al gt Page 35 F3 gt Page 62 DHCP IP assignment At power up starting endpoints search for a gt DHCP server and try to obtain configuration pa rameters from that central server The protocol is based on broadcasts and hence the endpoints do not need to know the location of the DHCP server DHCP is an extension of the BOOTP protocol The orig
23. cord clip Please make sure that such items are not accessible to children Never allow the telephone to come into contact with staining or corrosive liquids such as coffee tea juice or soft drinks The information provided in this document contains merely general descriptions or characteristics of performance features which in case of actual use do not always apply as described or which may change as a result of further development of the products An obligation to provide the respective performance features only exists if expressly agreed in the terms of contract Safety Precautions Impe Note for U S A and Canada only This equipment has been tested and found to comply with the limits for a Class B digital de vice pursuant to Part 15 of the FCC Rules These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a residential in stallation This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communications However there is no guarantee that interference will not occur in a particular installation If this equipment does cause harmful interference to radio or televi Sion reception which can be determined by turning the equipment off and on the user is en couraged to try to correct the interference by one or more of the following measures
24. download files from and to the OpenStage 5 SIP Value range table Permitted values numeric with DNS also alphanumeric Length max 15 digits incl dots with DNS also 92 digits Al gt Page 45 1 gt Page 65 Alphabetical Reference 95 DSS Address of Record Each DSS Direct Station Select will have the SIP Address Of Record AoR of DSS destination and will have an unshifted function key and LED DSS key assigned to it The assignment of key to DSS is determined by administration Value range table Permitted values numeric Length min 1 digit Length max 20 digits Al gt Page 43 5 gt Page 71 DSS Realm This field displays the realm of the DSS destination Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 43 gt Page 71 DSS user ID Enter the according SIP User ID of DSS destination Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 43 f gt Page 71 DSS password Enter the according SIP Password Value range table Permitted values alphanumeric Length min 6 digits Length max 24 digits Al gt Page 43 f gt Page 71 Emergency number Enter a valid emergency number Value range table Permitted values numeric Length max 20 digits A gt Page 40 f gt Page 63 96 Alphabetical Reference Feature Access This option allows the Administ
25. eee awe 1k wa V dee Rue dO Re ACE REA TE a s E A 3 Location of the Telephone 0 000 cece ete eee 4 Telephone Maintenance 0 0 tee eee 4 EAD EIS eee ee hee ease os eee ee dee eer cd a cette eh ooo N Gate ng aE Gps k ees eed oe 5 General Information 0 0 0 0 cee eee ee eee 8 About the Manual 0 000000000 ee ee ee ee 8 Intended US esse atc ak le ez ec e aces hE Mice itech cated Ran v edu dh th ade dade teas diy dec alti 9 Product Identification 0 6 esc es Se ke ee eee ae ela wae ee a aw ee ee a 9 Application Version zd eL ut REESE STRRERE ATUS EN ER RE eee deta a aa eee iets 9 SENICE pe C ot fae ane ocak ant Sapa ed bane Soe tunis a aeh ene aces ate 6 pare tes 9 Installation 2 lt 0 2 dons xk oed ooh oS tss ches Rew eee eee e 10 Prerequisites 0 000 ce ete eee eee 10 Connecting to the Network oee ssarrsnasddieneaaa eee 10 Installing the Phone 2 rss nidis attan niha ee eee eee pha 10 Power over LAN information i sea ere 0000000 ccc ce Rs 11 Startup Procedure ma aeai Ra E ete n 12 Using the OpenStage 5 SIP nananana 0 0 00 c eee eee 13 Dialling Keypad marres essaia c RR Ru E eA YR ERERE dd AXGGd EN ao 4 geared 13 Programmable Kays oade sedes ee bebe queen died d pIed irr q Cede ides 14 Control Keys e oone auran parmani nna naa ess rs 15 Phone FSSIU BS uuu us askus zRe6bx ER dod dom OR og ara x 16 Protocol support oa nc oca carac bisa iR AUORREREDAURRARAR
26. if the two lengths are equal or with CD1 Actually the timer can be set to values larger than the phone s interdigit timer it can be as high as 9 while the default internal timer is 6 Terminator A digit used to indicate that dialling is complete before reaching the maximum num ber of digits The terminator can only be or The terminator is sent as part of the digit string Option A special function to be applied when the digits are sent Currently two options are sup ported B Lock by pass Strings with this attribute can be dialled when the phone is locked other strings are barred by the phone E Emergency implies B Dialling these numbers will cancel both forwarding and DND to allow the emergency service to return calls to this caller There is nothing to prevent the user re invoking these features later Comment although this can be left blank it is useful to explain why this entry is present Field separators depend on exactly how the dial plan is put on the phone Raw database entries use the separator while an external document uses Dial plan entries are in priority order The phone will lock onto a matching entry and not check later ones except as described for the C action It is bad practice to have conflicting or duplicate entries as these may confuse other investigations Examples of Dial Plans Combinations of these examples can be used according to need These examples are delib
27. is prepared 1 or host name if DNS is applicable gt Page 24 56 Extended Administration Behaviour regarding the backup server Please make sure all parameters are set completely and accurately Backup Server not entered and activated Backup IP address or DNS name No IP address was entered gt Page 119 gt Page 80 Backup registration The Checkbox for the Backup registration feature is not marked Page 80 and or the feature was not activated in the telephone menu Page 119 The telephone only registers at the server OpenScape Voice In case the server fails or is not available some LEDs are blinking see Error Messages OpenStage 5 SIP page 130 Backup Server is entered but not activated Backup IP address or DNS name IP address is entered e g 192 168 1 1 gt Page 119 gt Page 80 Backup registration The Checkbox for the Backup registration feature is not marked Page 80 and or the feature was not activated in the telephone menu Page 119 The telephone only registers at the server OpenScape Voice In case the server fails or is not available some LEDs are blinking see Error Messages OpenStage 5 SIP page 130 Even after restarting the telephone no LEDs are blinking the telephone had registered at the OpenScape Voice However outbound calls are possible via the backup server while inbound calls to this telephone are not possible as it is not registered at the backup serv
28. methods Web Interface For remote configuration of individual IP phones in your network Direct access to the phone is not required Menu overview see Page 60 D To use this method the phone must first obtain IP connectivity The remote configuration is not applicable while the phone is not in idle mode Basic Administration 29 Basic Administration The phone is factory preconfigured to allow for a minimum of configuration activites required on the unit itself A number of parameters can be configured centrally by using a DHCP server When the phone is connected to the network it will react as follows If your network use a DHCP server the telephone will try to get its IP Address IP Address Mask SIP Addresses server gateway registrar SNTP Server Address Configuration Download Server Address and Time Offset from the DHCP server completely list see Page 22 In this case the telephone will boot with the IP address and will get the exact time from the con figured SNTP server You only have to configure the Terminal number SIP user ID and password Page 63 If the DHCP server is not available or configured to provide these parameters the telephone will become idle and has to be manually configured gt Page 30 Basic Configuration Configuration using DHCP Server The OpenStage 5 SIP is factory configured to have an IP address automatically assigned to it by the DHCP server as soon as it s connected to
29. nameplate containing the exact prod uct label and serial number on the bottom of the base unit gt Page 10 Please have these ready whenever you call our service department in case of trouble with or defects on the unit itself OpenStage 5 SIP 30817 S7400 A101 1 Ser Nr 0001 E320C244 Application Version To find out the current application version of your OpenStage 5 SIP see gt Page 51 Service The Unify service department can only help you with problems or defects on the telephone Ld unit itself Should you have any questions regarding the operation your specialist retailer or network administrator will gladly help you For any questions regarding the telephone connection please contact your network provider In the case of any trouble or defects on the telephone unit itself please dial the service number of your local distributor or your local Unify Branch office 10 Installation Installation Prerequisites The OpenStage 5 SIP acts as an endpoint client on an IP telephony network and has the following network requirements An Ethernet connection to a network with SIP clients and servers required A Dynamic Host Configuration Protocol DHCP server optional Either a Call Control System Proxy server There must be a device running RFC 3261 SIP compliant software Voice packet gateway optional Required if your VOIP Network is connected to the Public Switched Telephone Networ
30. of the loaded software The others relate to versions of internal software components Al gt Page 49 F3 gt Page 61 124 Alphabetical Reference VLAN discovery method Can be defined only if the gt Layer 2 support is activated gt Quality of Service QoS Use this function to define the location from where the gt Manual VLAN identifier should be fetched if 2 VLAN is used Manual The ID entered in gt Manual VLAN identifier is used DHCP If a gt DHCP server is used then the ID delivered by this server is ap plied Al gt Page 36 F3 gt Page 64 Voicemail number The number of where your voice mail server is located Value range table Permitted values numeric Length max 20 digits Either the Voicemail Number or the Message Waiting Address Page 105 should be entered but not both of them Al gt Page 40 F3 gt Page 63 Web Content Versionn Shows the version of the Web content of the OpenStage 5 SIP Al gt Page 51 gt Page 61 Alphabetical Reference 125 Abbreviations and Specialized Terms You will find more information in the relevant literature on the Network Technology and gt VoIP DHCP Abbreviation for Dynamic Host Configuration Protocol The DHCP is an Ethernet protocol that allows for the automatic configuration of IP based end points Additional information see gt Page 22 DNS Abbreviation for Domain Name System Additional information see Page 2
31. offer payload 100 to carry the DTMF events The far end may accept and confirm this payload or it may suggest a different payload val ue In this case the phone will follow that payload preferred by the far end On an incoming call the phone will follow the payload value suggested by the far end The phone is not capable of retrieving or understanding DTMF in band or DTMF in RTP informa tion it may receive This information is normally used by application or media servers to control feature access If the user presses keys when in a call connected state and in band DTMF has been negotiated he will hear the tones being sent in the speech path handset only If DTMF in RTP has been negotiated he will here clicks as speech packets are removed and replaced with DTMF in RTP key events See also gt Page 108 1 Server based feature de activated by access codes 18 Technical Overview Technical Overview Session Initiation Protocol SIP Overview The Session Initiation Protocol SIP is a ASCII based signalling protocol used for establishing sessions in an IP network A session could be a simple two way telephone call or it could be a collaborative multi media conference session Like other VoIP protocols SIP provides signaling and session management within a packet tele phony network Signaling allows call information to be carried across network boundaries Session management controls the attributes of an end to end call SIP wa
32. or abrasive powders Safety Precautions Labels Lid The device conforms to the EU guideline 1999 5 EG as attested by the CE mark This device has been manufactured in accordance with our certified environmental management system ISO 14001 This process ensures that energy consumption and the use of primary raw materials are kept to a minimum thus reducing waste pro duction All electrical and electronic products should be disposed of separately from the mu nicipal waste stream via designated collection facilities appointed by the government or the local authorities The correct disposal and separate collection of your old appliance will help prevent potential negative consequences for the environment and human health It is a pre condition for reuse and recycling of used electrical and electronic equipment For more detailed information about disposal of your old appliance please contact your city office waste disposal service the shop where you purchased the product or your sales representative The statements quoted above are only fully valid for equipment which is installed and sold in the countries of the European Union and is covered by the directive 2002 96 EC Countries outside the European Union may have other regulations regarding the disposal of electrical and electronic equipment Contents Contents Safety Precautions x 43 xd a4 CR PURA RA ICA HC HORE ede nd 3 Important Notes dua Desk ER a
33. pickup URI W gt Page 97 e SIP features Group pickup URI enter URI HotWarm Phone Phone type W gt Page 98 Default dial string W gt Page 98 e SIP features Phone typel select type Default dial string enter dial string Transfer on Ringing W gt Page 123 e SIP features Allow transfer on ringing mark to allow Callback URIs W gt Page 87 e SIP features Callback enter related access code Initial Digit Timer wW gt Page 98 e SIP features Initial digit timer set timer 1 to 120 40 Extended Administration Miscellaneous Web Interface path Parameter Menu Page 60 Emergency Number E W gt Page 95 SIP environment Emergency number enter emergency number Voicemail Number E W gt Page 124 SIP environment Voicemail number enter voicemail number Message Waiting Address E W gt Page 105 SIP environment Message Waiting IP Address or DNS name enter IP address 1 Either the Voicemail Number or the Message Waiting Address should be entered but not both of them Configuring Date and Time If the DHCP server in your network provides information about the SNTP server access the date and time is automatically shown on the phone If the DHCP server in your network does not provide a SNTP address you have to set the SNTP address manually If no SNTP server is in your network you have to configure the dat
34. starts on hook dialing There are four orig inating options A Keyset is assigned one of the following preferences Prime Line Preference The designated Prime Line is always selected for originating calls Idle Line Preference Any idle line is selected for originating calls with the lines selected based on line selection table for the device e g prime line first See also line rank on gt Page 103 and Terminating line preference on Page 122 Last Line Preference The line selected for originating calls is the line selected for the last call originating or terminating No Originating Line Preference A line key must be pre selected or post selected each time the user goes off hook Select the according line preference Idle line Primary Last None Al gt Page 42 F3 gt Page 68 Alphabetical Reference 107 Outbound proxy The OpenStage 5 SIP implements outbound proxy routing according to RFC 3261 If set the phone routes any request outside the context of an existing dialog to the configured proxy re gardless of the contents of the Request URI The phone does not apply this rule to requests sent within the context of an existing dialog These requests will always be sent to the address indi cated in the received Contact header the remote target or if present the Record Route header See also RFC 3261 If the user dials a URI p kelly dom1 com and the Outbound Proxy flag is Off and th
35. the DSS target 2 Make complete a call using the DSS target as the destination Completion of a call applies to cases where the user has performed an operation at the phone which results in them being prompted for destination digits DSS key configuration Each DSS key will be configured similarly to a Keyset line key and will require the following to be specified for the line SIP URI of the primary line at the DSS target SIP Realm SIP User ID SIP Password The remaining line configuration items will be forced to specific values for a DSS key line Parameter Web Interface path Menu gt Page 60 DSS e W gt Page 101 Funcktion keys Phone or Select a key with EDIT a configuration dialog appears Select DSS A key with DSS is ready for configuring DSS Address of Record E XW gt Page 95 DSS key configuration dialog Address of record enter e g phone number DSS Realm E W gt Page 95 DSS key configuration dialog Realm enter IP address DSS User ID E XW gt Page 95 DSS key configuration dialog User ID enter ID DSS Password E W gt Page 95 DSS key configuration dialog Password enter SIP password 44 Extended Administration Feature Access Web Interface path Parameter Menu Page 60 Feature Access e W gt Page 96 Feature Access mark as enabled if available Auto answer CTI Call hold explicit Call join Call transfer Do not disturb
36. the first 4 LEDs Lower Byte shows 1 the second 4 LEDs Step to the next character Upper Byte shows 3 the first 4 LEDs Lower Byte shows 1 the second 4 LEDs Step to the next character Upper Byte shows 3 the first 4 LEDs Lower Byte shows 2 the second 4 LEDs Step to the next character Upper Byte shows 2 the first 4 LEDs Lower Byte shows E the second 4 LEDs Step to the first character Upper Byte shows 3 the first 4 LEDs Lower Byte shows 1 the second 4 LEDs Step to the first character Upper Byte shows 3 the first 4 LEDs Lower Byte shows 8 the second 4 LEDs Step to the first character Upper Byte shows 3 the first 4 LEDs Lower Byte shows 6 the second 4 LEDs Step to the next character Upper Byte shows 2 the first 4 LEDs Lower Byte shows E the second 4 LEDs Step to the next character Upper Byte shows 3 the first 4 LEDs Lower Byte shows 1 the second 4 LEDs Basic Administration 33 Step to the next character Upper Byte shows 2 the first 4 LEDs Lower Byte shows E the second 4 LEDs Step to the next character Upper Byte shows 3 the first 4 LEDs Lower Byte shows 4 the second 4 LEDs Step to the next character Upper Byte shows 3 the first 4 LEDs Lower Byte shows 4 the second 4 LEDs Step to the next character All LEDs are off The IP address is complete and the operation has finished 34 Basic Administration After reboot the following parameters have to be configured
37. 123456 Terminate the operation You are now in the Administration Area The top two function keys take over the following functions in this area 1 Function key Press 1 function key to make settings 24 Function key Press 2 function key to view settings results ASCII codes see gt Page 81 Configure basics Press 1 function key Enter code Switch the DHCP IP assign off Terminate the operation Press 15t function key Enter code Enter IP address of the OpenStage 5 SIP to edit see gt Page 81 Terminate the operation Press 15t function key Enter code Enter terminal mask of the OpenStage 5 SIP to edit gt Page 81 Terminate the operation Press 1 function key Enter code Enter the default Route of the OpenStage 5 SIP to edit see gt Page 81 Terminate the operation Basic Administration 31 lt PROG gt DBO OOGO e lt PROG gt DBO zik e moe D o9 oc00 zt e moe 9D oo eo eo eo o eo eo eo o eo Only if you are working in a Virtual LAN VLAN Press 1 function key Input the code to define the manual configuration of the VLAN Discovery Mode Terminate the operation Press 1 function key Enter code Enter 0 4095 for the Virtual LAN ID Terminate the operation Confirm your entries and start the telephone Press key Enter the code Confirm the entry After restart you can make the other settings
38. 4 DTMF Abbreviation for Dual Tone Multi Frequence DLS The Deployment and Licensing Server DLS is a OpenScape Voice Management application that provides an integrated solution for the customers and the service personal to administer work points that are optiClients and OpenStage devices in OpenScape Voice and non OpenScape Voice networks EAP Extensible Authentication Protocol FTP Abbreviation for File Transfer Protocol Is used for transferring files in networks e g to update telephone software gt Download Applica tion G 711 Audio protocol for uncompressed voice transmission Requires a bandwidth of 64 kbit s 5 122 The G 722 recommendation describes ADPCM coding with a sub band The bandwidth for the sub band is 7 kHz at a sampling rate of 16 kHz The transfer rate is 64 kbps voice quality has a MOS rating of 4 5 which is quite high G 723 Audio protocol for compressed voice transmission The quality is worse than in gt G 711 and gt G 729 Requires a bandwidth of about 6 kbit s G 729 Audio protocol for compressed voice transmission The quality is worse than in gt G 711 and better than in gt G 723 Uses a bandwidth of about 8 kbit s 126 Alphabetical Reference Gateway Mediation components between two different network types e g IP network and ISDN net work HTTP Abbreviation for Hypertext Transfer Protocol Protocol for the transfer of data in gt IP networks
39. 4095 Default value 0 Al gt Page 36 F3 gt Page 64 Message Waiting IP address Use this function to configure the IP address or host name of the message waiting server Value range table Permitted values numeric with DNS also alphanumeric Length max 15 digits incl dots with DNS also 92 digits Either the Voicemail Number Page 124 or the Message Waiting Address should be entered but not both of them Al gt Page 40 F3 gt Page 63 NAT keep alive Is a mechanism of sending a periodic keep alive packet This prevent the Phone MAC Address from becoming aged out of the Switch security table and hence future packets will be forwarded as intended Value range is 10 to 3600 Al gt Page 35 1 gt Page 62 106 Alphabetical Reference Netboot filename Specify the name of the file containing the software of the OpenStage 5 SIP The file must exist in a defined directory on the gt FTP server gt Download server IP address or DNS name gt FTP path Value range table Permitted values alphanumeric Length max 92 digit Default value OS5nfli Al gt Page 46 1 gt Page 65 Netboot Versionn Shows the version of the Netboot of the OpenStage 5 SIP Al gt Page 51 gt Page 61 Originating line preference Originating Line selection provides connection of the phone to one of the lines appearing on it on an automatic basis when the user goes off hook or
40. 51 Display static Phone Information Parameter MAC Address wW gt Page 105 Web Interface path Menu gt Page 60 e General information MAC address Application Version W gt Page 123 General information Versions Application SIP Stack Version W gt Page 117 General information Versions SIP stack SIP Signalling Version W gt Page 117 General information Versions SIP signalling Web Content Version W gt Page 124 General information Versions Web content Netboot Version W gt Page 106 General information Versions Netboot Part Number W gt Page 107 General information Part Number Perform Diagnostic Tests Non user assisted diagnostic tests These types of diagnostic tests do not require assistance from a local user Parameter PING Test wW gt Page 108 Web Interface path Menu gt Page 60 Non user assisted diag tests Ping name IP address or DNS activate checkbox and enter IP ad dress RAM Test W gt Page 111 Non user assisted diag tests RAM test activate checkbox ROM Test W gt Page 113 Non user assisted diag tests ROM test activate checkbox 1 or host name if DNS is applicable gt Page 24 Extended Administration User assisted diagnostic tests These types of diagnostic tests require a local user to confirm the result at the OpenStage 5 SIP
41. Bit s ms Bytes Bytes G711 10 80 90 125 125 100 0 108 0 G711 20 160 170 205 215 82 0 86 0 G723 30 24 34 69 79 18 4 21 1 G 729 10 10 20 55 65 44 0 52 0 G 729 3 20 20 30 65 75 22 0 26 0 G 722 10 80 90 125 135 100 0 108 0 722 20 160 170 205 215 82 0 86 0 1 Inthe SIP environment it is possible to activate silence supression In this case the bandwidth calculation values are max values 2 For G 723 it is not possible to activate silence suppression and the only available sample size is 30 ms 3 In the SIP environment silence suppression is always active currently In this case the bandwidth calculation values are max values 4 For G 722 it is not possible to activate silence suppression Al gt Page 50 3 gt Page 66 Callback URIs For Callback Busy line and Delete callbacks enter the access code of the OpenScape Voice Callback No reply is for future use not supported with OpenScape Voice Al gt Page 39 F3 gt Page 64 Clear ALL user data This operation will clear out all personal data relating to the user including Personal Directory en tries and structure definition ready for the phone to be given to a different user It does not clear out Phone settings such as contrast or touch screen calibration settings nor Administration set tings such as network details A gt Page 54 1 gt Page 78 88 Alphabetical Reference Compression e
42. D Once a particular test has been started the local user will assume full control of the test us ing the keypad until the test is terminated Parameter Web Interface path Menu gt Page 60 LED test e XW gt Page 101 po me diagnostic tests Key test e l l l W gt Page 99 User assisted diagnostic tests Key test Audio loop test W gt Page 84 User assisted diagnostic tests Audio loop test Line monitor PE W gt Page 102 User assisted diagnostic tests Line monitor Security settings Parameter Web Interface path Menu gt Page 60 Payload Security allowed e W gt Page 108 Security Settings Payload Security allowed mark as allowed Connectivity check interval E W gt Page 88 Security Settings Connectivity check interval enter value SIP Server y Validate Security Settings XW gt Page 116 SIP server validation mark as enabled Restart the OpenStage 5 SIP Restart by software Precondition A confirm key is established Page 69 Web Interface path Parameter Menu gt Page 60 Restart Terminal press successively confirm with confirm key Restart terminal the connection to the phone will be lost temporarily Extended Administration 53 Restart by Hardware In the case of PoE please remove the network connection In the case of external power supply please remove the power supply only Restore Factory Setti
43. EA HA ERRARE RE ELE 16 CapabilitiGS 2c Reg aea ae Pe to Re UR a d CA Re Tx c RR RC Ga UA dnce t BL 16 Call Fealt les oe o EU ed Beis Bg Pie BS Pes ih a ru due te a hk 17 DIME Lesna pict cee at Nene tee nag ae ab ares a natant ate da ANG abet ot ane note ae a ane 17 Technical Overview lees 18 Session Initiation Protocol SIP 2 0 0 cee 18 IP Network Protocols rs Bea bed baie ead Baad wae Oe eeu eas ed I 22 IP Network Configuration 0 0 0 0 cee yr 26 Administration Interfaces 0 000 c eee ee ee 28 Webs lntemace boe eer ce Be Os a a a oe ee Ee RU CE 28 Basic Administration 2 0000 eee eee eee 29 Basic Configuration 0 0 cette ee 29 Contents 7 Extended Administration 22222 2r s 39 Configure Network Parameters llis hn 35 Configure System Information 0 0 0 0 n 37 Configuring Date and Time 1 6 1 hm ms 40 MIDI CREE p 41 Dial Plan Configuration and Status 0 0 0 cee ees 42 Direct Station Select DSS 0 0 0 00 ce teens 43 Feature ACCeSS esate ota pea ei ee Dee 24 dA was Se earn eae pare 44 Function Key assignments 0 20 0000 eee 44 Software Update Transferring Files 0 0 00 teens 44 Port Numbarlng cre Rec tm en KEENE ee n mae I irn rade we dl BR a be 47 Configuration Management 00 0c ete ee 48 Use SNMP censendus Seana anaes DUM EAA tng XO Rus scudo E R aan yee 49 Change Speech Parameters pocs reer 000
44. GPU New Call Beep Message waiting Music on hold Hot keypad dialing Callback busy Callback no reply Function Key assignments Parameter Web Interface path Menu gt Page 60 Function key e W gt Page 97 Een keys Phone Select a key with EDIT a configuration dialog appears Select a function in the list configure parameters in the dialog if required Software Update Transferring Files The OpenStage 5 SIP is capable of transferring files using the gt FTP protocol This feature can be used to update the phone software and up or download the phone s configuration file The phone acts as a FTP client and requires a FTP server in the IP network where the files are located or can be placed Application Software Update If itis necessary to change or upgrade the application software of your OpenStage 5 SIP perform the following Find out the current application version of your OpenStage 5 SIP Page 49 Decide whether an update is useful and necessary ez Be careful Consider that the software must be compatible with the telephone If useful download it from Software Supply Server SWS and install the application software via FTP gt Page 46 Extended Administration 45 FTP Server Requirements There are no specific requirements on the functionality of the FTP server to be used Any FTP server providing standard functionality will do There is a variety of servers available includin
45. LAN Impe The Authentication is done via digital Certificates For detailed informationen refer to the IEEE 802 1x Configuration Management Administration Manual If your network use a DHCP server the telephone will try to get its IP Address IP Address Mask SIP Addresses server gateway registrar SNTP Server Address Configuration Download Server Address and Time Offset from the DHCP server completely list see Page 22 In this case the following parameters have to be configured using the Web Interface Administra tor Menu Parameter Web Interface path Menu gt Page 60 Terminal Number E W gt Page 121 SIP environment ij Terminal number enter terminal number SIP User ID e W gt Page 117 SIP environment SIP user ID enter ID SIP Password y W gt Page 115 SIP environment New Confirm SIP password enter re enter password 30 Basic Administration 000 or 2 00 i moe D mw ep moe p oo o eo moe 9e oo x eo moe gt oo Ei eo moe ep oo eu e Manual Configuration If your network does not use a DHCP server you must disable the DHCP IP as signment manually and specify the phone s IP address and subnet mask and the network gateway IP address default route for the phone Entering the administration area Press the keys simultaneously Press the keys successively all LEDs flash Enter admin password default
46. N Port Settings Only available in Web Interface FPN Port Settings WARNING If you make changes to the fields marked with an asterisk you will have to restart the terminal manually before they take effect a e Bxn FM s0 co soo so mo CEN O EINE Reset J E 80 Web Interface Survivability Resilience and Survivability 2 Page 54 Survivability Web Interface 81 OpenStage 5 SIP IP number editor The IP number editor permits you to enter a standard IP address of the form w x y z including the dots Note that leading zeros are ignored therefore you may omit them and the editor will over write any pre existing number In order to insert the dots between the digit fields of the IP number you should press the phone s hard key ASCII Result Codes e e e e e e e e e e e e e e e e e e e e e e e e e e e e e o e LJ o o e e e e e e e e o e LJ o e e e e e e e e e e e e e e e e e e e e e o o S e e e e e e e e e e e e e e e e e e e e e e e o e e end dot 0 T 2 3 4 5 T 8 9 0x00 Ox2E 0x30 0x31 0x32 Ox33 0x34 0x35 Ox36 Ox37 0x38 0x39 0 46 48 49 50 51 52 53 54 55 56 57 82 Alphabetical Reference Alphabetical Reference This reference offers basic information that can be used by the administrator to carry out admin istration and diagnostics related jobs in the Op
47. P environment gt File transfer gt Time and date gt SNMP gt Non user assisted tests General information A Display static Phone Information gt Page 51 Display Application Version gt Page 49 Display Upload Download Status gt Page 48 Dial Plan Configuration and Status gt Page 42 General Information MAC address 00 01 e3 24 77 e8 Versions Application V3 R0 13 53 SIP stack 4 0 28 28 SIP signaling 0 0 1 Web content V3 R0 13 53 Netboot 2 01 Part Number S30817 S7201 A101 12 Dial plan name amp status None File Transfer status Last transfer Status Application file download 07 57 21 Apr 2011 OK Configuration file download never Configuration file upload never Hold music file download never System configuration download never Phone configuration download never Personal directory import never Personal directory export never 62 Web Interface Network IP and routing Network Addresses Page 35 Network IP Address and Routing 192 168 1 251 0 0 0 0 0 0 0 0 Web Interface 63 SIP environment Configure System Information gt Page 37 AN WARNING If you make changes to the fields marked with an asterisk you will have to restart the terminal manually before they take effect 4989700731285 4989700731285 ift oscft global intra 0 ift oscft global intra 0 iftbr oscft 0 5010 64 Web Interface
48. P related traffic for a PC Prioritization of voice traffic over that of the HTTP traffic means that during periods of heavy network load that voice service is maintained whereas the response times for a user s Web Broweser will degrade and possibly stop working Quality of Service can be supported in networks at both Layer 2 and Layer 3 At Ethernet layer 2 the MAC header is extended to provide VLAN information and Quality of Service priorities Ether net layer 2 allows for prioritisation of traffic from O lowest to 7 highest At the layer 3 the IP layer traffic can be prioritized using information embedded in the IP Type of service DiffServ field that allows for 64 levels of prioritization To utilize Quality of Service features the network infrastructure switch fabric must support prior itized delivery of traffic based on layer 3 and or layer 2 Secure Payload OpenStage 5 SIP telephones enable you to establish a secure telephone connection provided that the recipient s telephone is also capable of this Voice transmission is encrypted and subse quently decrypted by the called party s telephone and vice versa Even the signaling for the call setup and the exchange of the encryption data is carried out via a secure connection The tele phones have to have a valid registration at an SIP server via TLS 28 Administration Interfaces Administration Interfaces You can configure the OpenStage 5 SIP by using any of the following
49. a Web Client such as Internet Explorer Registration a Note that registration only occurs when the SIP Routing mode Page 116 is setto Server Registration is the process by which centralized SIP Server Registrars become aware of the ex istence and readiness of an endpoint to make and receive calls The phone supports a number of configuration parameters to allow this to happen Registration can be authenticated or un authenticated depending on how the server and phone is configured For unauthenticated registration the following parameters must be set on the phone Terminal number Page 121 or Terminal name Page 121 SIP Routing Page 116 set to Server SIP Server Registrar address gt Page 114 configured IP address or host name In this mode the server must pre authenticate the user This procedure is server specific and is not described here The phone supports the Digest authentication scheme and requires the following parameters to be configured in addition to those for unauthenticated registration SIP user ID gt Page 117 SIP Password gt Page 115 SIP Realm optional gt Page 115 For authentication to work the server must have created an account for the user with matching user ID password and Realm parameters Tl Note a challenge from the server for authentication information is not only restricted to the REGISTER message but can also occur in response to other SIP messages eg INVITE Bel
50. a line remains reserved for a user who is dialling after this timeframe another user who s phone is using the same line can access the line A gt Page 42 3 gt Page 68 Alphabetical Reference 113 Ringer Settings See Alert indication Page 83 Rollover type The Rollover ring setting will be used if the Keyset is currently active in a call when an incoming call arrives on a different line Selectable rollvover types are 1 No ring 2 Alert ring 3 Standard ring 4 Alert beep A Page 42 1 gt Page 68 Rollover Volume While you are active on one line of a keyset telephone the rollover ringing feature signals addi tional incoming calls on other lines The volume can be set from 1 to 5 For more information see operating manual Al gt Page 42 F3 gt Page 68 ROM test Use this function to test the gt ROM memory of your OpenStage 5 SIP The results are displayed after the test Al gt Page 51 1 gt Page 74 RTP packet size Use this function to define the RTP G711 G729 packet size Options are Auto recognition 10 milliseconds 20 milliseconds Al gt Page 50 F3 gt Page 66 Secondary DNS IP address See Primary DNS IP address Send Generic Traps to Management Center Allows the user to control whether or not the phone sends generic standardised traps to the man agement center see gt Page 118 gt Page 66 114 Alphabetical Reference Send QDC Traps to M
51. a port and transmits this together with the IP address to the telephone Usually both sides use a port higher than 1023 This method is used if the server cannot reach the telephone This is for example the case if the telephone is switched by a router which translates the telephone address using NAT or if a firewall shields the network of the telephone from external attacks Al gt Page 45 1 gt Page 65 Alphabetical Reference 97 FTP password Enter the password defined in the gt FTP server as password for accessing this server The password must correspond to the gt FTP username and match the password on the server Value range table Permitted values alphanumeric Length max 24 digits Default value 123abc After factory reset the FTP password is replaced with the string 123abc Al gt Page 45 F3 gt Page 65 FTP path Enter the path of the directory defined in the gt FTP server for uploading and downloading files Value range table Permitted values alphanumeric Length max 92 digits Default value Al gt Page 45 1 gt Page 65 FTP username Enter the name defined in the gt FTP server as user for accessing the server The password must correspond to the FTP password and match the username on the server Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 45 1 gt Page 65 Function key A key on the OpenStage 5 SIP
52. aditional public emergency number 99 S 8 Nottingham local numbers 98 S 8 Nottingham local numbers 9112 S 4 E European standard emergency number 92 Alphabetical Reference 9118 S 6 Directory enquiries although this would often be barred by a PABX 91XX C 4 4 3 Special services in theory longer numbers can be used but there are none at present Use of C will allow these with changes to this plan 900 S 13 20 3 International numbers 906 8 11 12 3 National numbers Secondary dial tone might be wanted after the initial 9 In the initial case this can be done by changing the action from S to SD1 When the more refined set of numbers are used they would need to be preceded by 9 CD1 1 1 give dial tone after 9 Feature codes starting with and might be useful Studies similar to that for the public numbers would be needed but standard values might be possible for a given server A private network replacement would need longer local numbers possibly with the leading digits being used to nominally identify the site Examples would add nothing to those above British public network usage would need entries similar to the ones used to access the public net work above but removing the leading 9 the access code and having one less digit More dis crimination on the international numbers might be used The American public network uses a different technique for discrimina
53. alm ID is required to be registered with the line key see SIP realm gt Page 115 SIP password gt Page 115 and SIP user ID gt Page 117 130 Troubleshooting LEDs on the phone Ls ww lt v vv vvv ofw y v v v v v w lt v vvv v EDE v v v v v w Zu v v v v Error Messages OpenStage 5 SIP The LEDs flash till you switch to the Administration Mode gt Page 30 No IP address The DHCP server cannot assign a terminal IP address gt Page 29 Code 1 Possible solution Check the DHCP server Terminal Mask not assigned The DHCP server has failed to assign a Terminal Mask gt Page 29 Code 2 Possible solution Check the DHCP server No Default Route The DHCP server cannot assign a default route Page 29 Code 3 Possible solution Check the DHCP server No IP Address is set The DHCP IP assignment Page 30 is switched off and no terminal IP address is configured Code 4 Possible solution Activate the DHCP IP assignment Page 30 Enter the terminal IP adress gt Page 30 No terminal Mask is set The DHCP IP assignment gt Page 30 is switched off and no terminal Mask is configured Code 5 Possible solution Activate the DHCP IP assignment gt Page 30 Enter the terminal Mask Page 30 No Default Route is set The DHCP IP assignment gt Page 30 is switched off and no default route is configured Code 6 Possible solution Activate the DHCP IP assignment gt Page 30 Enter the default
54. an in coming line is selected based on its ringing assignment Ringing lines are selected on a first in first out basis first then alerting visual only lines are selected on a first in first out basis Incoming Line Preference with prime line preferred Same as Incoming Line Preference but if the prime line is ringing at any time it is signaled and selected before calls on secondary lines Prime line must have ringing arrangement No Terminating Line Preference A line key must be pre selected or post selected each time the user elects to answer a call Select the according terminating line preference Ringing Incoming Incoming PLP Ringing PLP None See also Originating line preference on gt Page 106 Al gt Page 42 gt Page 68 Time zone offset The specification describes the shift in hours corresponding to the time zone information of the SNTP server Make an entry only if an gt SNTP server provides time zone information Value range table Permitted values numeric min 12 max 12 Default value 0 Al gt Page 40 F3 gt Page 65 Transaction timer This timer is timeout when phone uses the 2nd destination address from the DNS SRV query The default is set to 32000msec 32sec and can be changed to minimum value of 3000msec 3sec If a phone uses this mechanism one time it will remember the 2nd destination as the primary one for a time of 10 minutes After that i
55. anagement Center Allows the user to control whether or not the phone sends QCD Quality Data Collection traps to the management center see Page 118 gt Page 66 Show focus The Show focus option allows a Keyset to be set so that the LED of the line that is currently being shown in the display flutters to identify it Al gt Page 42 53 gt Page 68 Silence Suppression It suppresses transmission of packets on no conversation Effects the following codecs G 711 G 723 und G 729 Al gt Page 50 F3 gt Page 66 SIP addresses Use this function to define the following IP addresses or host names IP address SIP Server IP address or host name and port of the SIP proxy server OpenScape Voice SIP Registrar This field is only used when the phone is in Server routing mode gt Page 116 It contains the IP address or host name and port of the reg istration server to which the phone will send REGISTER messages Ei ther an IP address or a host name may be entered With an address en tered in the SIP Registrar field the phone will register and be able to receive incoming calls but in order to make outgoing calls it is also nec essary to enter an address into the SIP Server field see above SIP Gateway and IP address or host name and port of the SIP gateway E g for a hardware Port box to phone directly into the public network conversion of SIP to TDM SIP routing has to be set to Gateway for this fu
56. at 580 Not configurable DNS 53 DHCP Server port 67 Default BOOTP port num bers Not configurable DHCP Client port 68 Default BOOTP port num bers Not configurable FTP 21 TFTP 69 Not configurable Service Agent Request Port 5100 SA port base Default value 5100 Auto discovery 5100 SA port base 0 Config Service 5130 5130 SA port base 30 QDC server 12010 12010 DLS 18443 18443 Sip server 5060 5060 he port numbers and port base numbers will be configurable via the Web pages and by the DLS Note that changing the value of a port number may require the phone to restart For detailed information please use the IFMDB 48 Extended Administration Configuration Management Specify configuration update file Parameter Configuration update Dis DL Params E Page 88 gt Page 88 Web Interface path Menu gt Page 59 e Configuration management Settings Deployment Service DLS IP address or DNS name Port Configuration update Management Tpye Not in use e Configuration management Settings Secure configuration download HTTPS Use secure configuration download not in use Configuration update Management Tpye Not in use e Configuration management Settings Non secure configuration download FTP Use non secure configuration download not in use Display Upload Download Status Before you transfer a file it could be useful to have a loo
57. ate option 255 have to close the option frame Example Code 4 length 28 data sdlp dis siemens com 18443 22 2 22 22222333 333 33 3 3 4 4 4 4 4 4 4 4 4 4 5 5 5 5 5 55 012345 67890 12345678 90123456 789012 3414 c Option 1 data dls server address OO a oo O B ocoooonsni soc ac O olo o o o OID coo O 1 a ja n O 312s d P p PPP Vd i s sipe elr S Cel o PELE EZIO JOO 0 0 25 8p oy m m 5 The five Padding Bytes and the terminate option 255 now completes the option frame in byte 48 Using Vendor Classes A Vendor Class is used to make sure that vendor related information is only sent to the tele phones instead of sending it to all other terminal devices as well By using a vendor class vendor information elements for each vendor class can be sent to all devices of this vendor class The vendor class name is OptilpPhone OpenStage 5 SIP telephones send their vendor class name using the option 60 to the DHCP server whenever they request data from the DHCP server If the option VLAN Discovery is set to DHCP on the telephone the telephone registers using the vendor class name OptilpPhone during the initial boot process and then using the vendor class name OptilpPhone On the DHCP server you can therefore use the vendor class name OpenStage 5 SIP to assign the VLAN and use the vendor class name Optil
58. ate the warranty and the CE mark Use only original accessories Using other accessories may be dangerous and will in System Support A Never open the telephone or a key module If you encounter any problems contact Attention If the OpenStage 5 SIP is supplied with power over the LAN interface gt Page 127 the power source must be a limited power source PowerHub compliant with IEC 60950 This equipment has been tested and found to comply with the limits for a Class B digital device pursuant to Part 15 of the FCC Rules These limits are designed to provide reasonable protec tion against harmful interference when the equipment is operated in a commercial environment This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communica tions Operation of this equipment in a residential area is likely to cause harmful interference in which case the user will be required to correct the interference at his own expense The IP telephone OpenStage 5 SIP complies with the European standard EN 60 950 The earpiece in this telephone handset contains a magnet To prevent injury before each use ensure objects such as pins or staples are not stuck to the earpiece There is always the danger of small objects being swallowed by young children In the case of the OpenStage 5 SIP this applies in particular to the connecting
59. atible such that only one end of a call need understand the draft and implement the SIP extensions for it to work Default value Off This is because some server environments support their own mechanism for auditing the health of a session Al gt Page 38 F3 gt Page 63 SIP Signalling Version Shows the version of SIP signalling of the OpenStage 5 SIP Al gt Page 51 1 gt Page 61 SIP Stack Version Shows the version of the SIP stack of the OpenStage 5 SIP Al gt Page 51 gt Page 61 SIP Transport Use this function to define the transport protocol Protocol Use gt UDP for SIP messages Use gt TCP for SIP messages Use gt TLS for SIP messages Al gt Page 38 F3 gt Page 63 SIP user ID User name Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 38 F3 gt Page 63 118 Alphabetical Reference SNMP MIB2 errors Lists the following packets Discarded inbound packets Invalid inbound packets Discarded outbount packets Invalid outbound packets gt Page 66 SNMP password Specify the password that was defined in the gt SNMP server as the password for accessing this server Value range table Permitted values alphanumeric Length max 24 digits Default value public Al gt Page 49 F3 gt Page 66 SNMP Queries Allowed Allows the user to control whether or not the phone responds to gt SNMP qu
60. ay be active at a time connected to the handset or speaker mic but there may be several other calls connected to keyset that have been held or are alerting the Keyset The line keys may be used to select different lines A keyset LED for a shared or private line reflects the status of a call on the line whether the call is connected to the keyset or to another keyset with an appearance of the shared line Select the desired Type Shared Private Al Page 41 53 gt Page 70 104 Alphabetical Reference Line password Enter the according SIP Password Value range table Permitted values alphanumeric Length min 6 digits Length max 24 digits A gt Page 41 1 gt Page 70 Line Primary line Every keyset has a primary line with an Address of Record that typically represents the user of the keyset line owner Only one line on a keyset can be designated as the primary line by the downloaded configuration all other lines are secondary or and referred to as non primary lines Impe A non Keyset phone only has a single line which is considered to be the same as the primary line of a Keyset Both primary and non primary lines can be shared or private For example a shared primary line appears as a secondary line on another keyset which is then able to monitor the calls to the own er of the primary line Some features can only be applied to the primary line to ensure that conflicting feature s
61. cee eee eee he 50 Configure Ringer Settings 0 illie mn 50 Display static Phone Information 0 000 ccc eee 51 Perform Diagnostic Tests oneni cirrerierr esdi iar ee eee 51 Security SONGE 0 222640 de dad didi a E adele date dates da UIRGO dda Sade ed s 52 Restart the OpenStage 5SIP 0 1 teas 52 Restore Factory Settings llle 53 Change Administrator Password liie nn 53 Reset User Password iil er Gs Leu eue dee Ed Rd e pud PER Pd 53 Clear ALL user data ll llesulleeeleeeel ell hh hne 54 uei EP 54 Resilience and Survivability ssi I 54 SIP Security Configuration 0 0 cent en 57 WED IMCIIACG s c 24 nein es EGGUOE ERR RR ERRARE Ed 59 Establishing the Connection to the Phone 00 0 c cee tenes 59 Web Pag6S 2 estt te ed eoe Re Wee ene beta ee a we BSW em RR RUM RE Ae POR Re Dn RD 61 OpenStage 5 SIP IP number editor lilii 81 Alphabetical Reference 00 ccc eee ees 82 Description of Functions 0 000 cece sr 82 Abbreviations and Specialized Terms 000 00 cee eee eens 125 MOUDISSNOOUIEI i23 29 arsaa 9 dod eee ood oe CROIRE heme 129 General Troubleshooting Tips illie 129 Fault Finditlg i22 beRRDTeWLOUpbeniebebppde ddev eed qup pude iter s 129 Phone Configurations ie a x53 go o 9 dopo asi ma nia EAR en 133 Common Configuration Factory Defaults 0 0 00 eese 133 Product support on the internet llle es 134
62. compatible Speech Codec settings to avoid Duplex Missmatch Page 84 Ensure that the Room Character is correctly configured for the type of room the phone is located in see User Manual chapter Room Character Phone Configurations 133 Phone Configurations This section identifies the configuration settings to allow the phone to operate in OpenScape Voice environments This configuration is a common one dealing with settings generic to all systems System specific ones follow Common Configuration Factory Defaults Table 1 Basics Function standard value Administration password 123456 DHCP IP assign on LAN Port Setting Auto VLAN Discovery DHCP QoS L2 L3 On On SNMP password public Time zone offset 0 User password 000000 134 Phone Configurations Table 2 Extended Product support on the internet WEB page Parameter SIP details SIP port RTP Base port SIP transport SIP server type SIP session timer value Registration timer value Transaction timer Registration backoff timer SIP features Auto answer Beep on auto answer Auto reconnect Beep on auto reconnect Allow transfer on ringing Initial digit timer Quality of Service QoS Required Voice Signaling Default Required Voice Signaling VLAN Discovery Codec Audio mode RTP packet size Silence Suppression Play DTMF RFC
63. d This timer will only be applied if there is no response timeout an error answer 500 503 without RETRY AFTER header or any other error response next to 403 500 503 Al gt Page 38 F3 gt Page 63 Registration LEDs This option determines whether the line LEDs will be lit to show if they have been registered suc cessfully when the phone starts up If set to be On then as each line is successfully registered its LED will be set ON Al gt Page 42 1 gt Page 68 112 Alphabetical Reference Registration timer value This field determines whether the phone sends an expires header in the REGISTER messages that it sends and if so to what value it sets it The expires header in a REGISTER is a suggestion to the Registrar server of how long it should be before the phones registration expires To stop its registration from expiring the phone has to send another REGISTER to the Registrar before its current one has expired The expires value which the phone sends is only a suggestion the actual value to be used will be supplied to the phone by the Registrar in the OK message that it sends in response to the REG ISTER Normally this will be the same as the value that the phone has suggested but if the sug gested value is outside the Registrars range of acceptable values then it could be different The phone actually adds 20 to the value that is puts into the expires header so that if the REG ISTERs that it sends get delayed b
64. d by DNS SRV set the SIP registrar server port to 0 too The web inter face path is SIP Environment gt SIP details gt SIP registrar server As SIP gateway address enter the DNS domain name for which the DNS SRV records are valid The web interface path is SIP Environment gt SIP details gt SIP gateway address Additionally if the SIP server specified in SIP Environment gt SIP registrar server address is to be configured by DNS SRV set the mentioned parmeter to the DNS domain name for which the DNS SRV records are valid Additionally the transaction timer gt Page 122 and if TLS is used the Connectivity Check Timer gt Page 88 have to be configured In survivability mode some features will presumably not be available e g callback Extended Administration 55 Survivability with a proxy A survivability proxy acts as a relay between the phone and the primary SIP server Thus the ad dress of the survivability proxy is specified as gateway at the phone When the connection between the survivability proxy and the SIP server breaks down e g be cause of server failure the survivable proxy itself acts as a replacement for the primary SIP server Vice versa in case the phone can not reach the survivability proxy itself it will register with the primary SIP server provided in the DNS SRV server list The survivability proxy notifies the phone whenever the survivability changes Furthermore to en hance survivabilit
65. data is downloaded e g device name If the option Use E164 as hostname was activated for a basic user the E164 number of the basic user is entered to the Terminal Hostname field when data is downloaded Overview of the HostName handling on normal operation mode Host Name Result Change Hostame Change E164 Change Hostame Change E164 Al gt Page 35 1 gt Page 62 Alphabetical Reference 121 Terminal IP address Enter the gt IP address for the OpenStage 5 SIP if not provided by gt DHCP dynamically gt DHCP IP assignment f the value was assigned dynamically it can only be read The change will only have effect if you restart the phone Value range table Permitted values numeric Length max 15 digits incl dots Al gt Page 35 F3 gt Page 62 Terminal mask Enter the gt Subnet Mask for the OpenStage 5 SIP if not provided by gt DHCP dynamically gt DHCP IP assignment f the value was assigned dynamically it can only be read The change will only have effect if you restart the phone Value range table Permitted values numeric Length max 15 digits incl dots Al gt Page 35 gt Page 62 Terminal name The phone will send REGISTER messages containing the contents of this parameter but only when the Register by Name field gt Page 111 is set to On and the SIP Routing field 2 Page 116 is set to Server Wh
66. dation If this function is activated a server certificate is requested and validated during the OpenScape Voice registration A gt Page 52 gt Page 77 SIP session timer value The expiration time for the session is set via the system SIP Session Time value This value is ignored if the SIP Session Timer is not enabled The RFC standard defines a Minimum expiry time which is 90 sec For detailed information see RFC 4028 paragraph 4 Value range table Permitted values numeric Range 90 3600 seconds recommended 1800 Default value 3600 seconds Al gt Page 38 F3 gt Page 63 Alphabetical Reference 117 SIP session timer enabled The phone supports the SIP draft ietf sip session timer 08 For detailed information relating to this draft please see http www ietf org Session timers provide a basic keep alive mechanism between 2 user agents or phones This mechanism can be useful to the endpoints concerned or for stateful proxies to determine that a session is still alive This is achieved by the phone sending periodic re INVITEs to keep the session alive The inter val for the re INVITEs is determined through a negotiation mechanism defined in the above draft If a re INVITE is not received before the interval passes the session is considered terminated Both phones are supposed to terminate the call and stateful proxies can remove any state for the call This feature is sufficiently backward comp
67. e e W gt Page 96 File transfer z Use Passive Mode FTP mark to use 1 or host name if DNS is applicable gt Page 24 46 Extended Administration Upload Configuration File The OpenStage 5 SIP allows you to upload the phones configuration file Uploading will be done in the ASCII format to the FTP download server with its common settings The default name of the file which will be uploaded is os5c without an extension Regard that the system password is encrypted Parameter Web Interface path Menu Page 60 Upload Configuration E W gt Page 123 File transfer submit with upload configuration Upload configuration Downloading Files Specify and download application file Parameter Web Interface path Menu gt Page 60 Application Download filename W gt Page 84 File transfer Download Application Appicanon mename enter filename W gt Page 94 Action on submit select Download type or Download Application IIE After the download is completed it will cause the OpenStage 5 SIP to restart Specify and download configuration file Parameter Web Interface path Menu gt Page 60 Configuration Download E Filename File transfer W gt Page 88 Configuration filename enter filename Action on submit select Download configuration Download Configuration W gt Page 94 i Upload configuration Specify and download netboot file
68. e eee tat nae 13 14 Q QoS Quality of Service L Overview een 27 Labeling key fields eseseeee 13 LAN port settings seeeessseesese 100 R LAN port speed ceceeeeeeeeceeceeeeeeeeeeeeseeneenneateees 100 Last restart eed e Bee neis 101 RAM IESI ascteetundiundidittsuutonddiegi eee 111 Layer 2 Default ssssssseeene 100 baud by Da INED n Layer 3 signalling eeeeeeeesssss 100 Pegiawangn EEDS satio tsi pine terete T Layer 3 voice ssssssssee eee 100 Regie alot procedura Bid Ce be ecg etty we Oe Ope a LED IOSU enat anen denen mec entes 101 Registration timer value s cio uabeciestesb An sdgtissi diente 112 LEDs light emitting diodes 12011002 13 Isollover type uiii o e ema ie 113 Lina Rollover Volume sss 113 Address ot BIB ee cse e Eie a 103 ROM leSU iu edt eiie dtes eee perd edis eed et cias 113 Hunt Ranking n 103 Routing SEW coo od face tke onsale ceesuahiescs 104 IP oa 99 Primary line aoaiina 104 oM a a 26 e eer A ANDR 44 RE Ringe a a a i d uei 104 PACKON SIRE tedious t biis oneitfetois i User ID ssssssssseeeenenneenes 95 104 Line key operation mode sseeeeeeee 102 Line monitor 102 S Location of the Telephone ssee Serial NUM E n n 9 Shared type edt etate texte i en ud ed n a 103 Silence suppression
69. e Server Registrar is in domain dom2 com the phone will attempt to resolve the domain part of the URI dom1 com the result will be form the request URI that is sent to the server in the dom2 com domain If the user dials a URI p kelly dom1 com and the Outbound Proxy flag is on and the Server Registrar is in domain dom2 com the phone will not attempt to resolve the domain part of the URI dom1 com the request URI will contain p kelly dom1 com but will be sent to the Server Registrar in the dom2 com domain It is then upto the Server Registrar to determine the loca tion of the dom1 com domain and forward the request there With the Outbound Proxy flag Off if the user types just a phone number or name the domain part is automatically per pended and is based on the configured Server Registrar domain name or IP address eg If the Server Registar is in dom1 com domain and the user dials 123456 the request URI will be 123456 dom1 com The phone will resolve the dom1 com part via DNS and forward the request there With the Outbound Proxy flag On if the user types just a phone number or name the domain part can come from 1 of 2 possible sources If the menu item Default OBP domain gt Page 89 is configured then this will be per pended to the name or number If it is not configured then the Server Registrar domain will be per pended and the request sent If the Outbound Proxy flag is On and the Default OBP domain is set and th
70. e and time manually Ime If SNTP is being used any user specified value for Time and Date will be overwritten when the next SNTP update occurs SNTP is available but no automatic access by DHCP server Parameter Web Interface path Menu Page 60 SNTP Address p W gt Page 118 Time and date SNTP server IP address or DNS name enter IP address Timezone Offset E W gt Page 122 Time and date Time zone offset enter timezone offset 1 or host name if DNS is applicable gt Page 24 No SNTP server available Web Interface path Parameter Menu gt Page 60 Date Time c W gt Page 88 Time and date Local time Date enter Local time enter select Date Daylight Saving Time E XW gt Page 89 Time and date z Daylight saving activate deactivate checkbox Extended Administration 41 Multiline Line key configuration In line overview menu you can configure lines and an assign lines to keys It suffices to assign one line to a key for to go in multiline operation Parameter Line W gt Page 101 Web Interface path Menu gt Page 60 Funcktion keys Phone Select a key with EDIT a configuration dialog appears Selectiline A key with line is ready for configuring Address of record W gt Page 103 Line key configuration dialog Address of record enter e g phone number Realm wW gt Page 104 Line key configuration dialog Realm ent
71. e dee sta n nies 14 BIDS 88 Date configuration aiseria 40 Daylight saving eee 89 Default domain name cccccceessecceceeeceeeeeeeeeeaaees 89 Default gateway sse 89 Default host name 2 0 0 cece ccccceecesseeceeceaneeseeeeeeeaaees 22 Default Route eesseseeeseeme 30 Default route enini a ai 89 G G 711 Silence Suppression ssssss 114 G711 eI c 84 Silence suppression sssssssssss 114 138 Index G723 COETS m a 84 N G729 Namieplate n eere enel t a a 9 Gio eam 84 Group pickup URI sesssssseee 97 O OBP domal 5 ottiene venti tont des 107 Originating line preference sessssss 106 Interfaces for administration sesess 28 Outbound proxy cuuududscnacidiscinritl deii eiua dei vd Mini 107 Invalid in outbound packets sssssss 99 IP TOULIDIQ as ocior entre rti peti rete any 99 p K PING c 108 Ping test eeeeesesssesseeeeesenennee entente nnn 108 Key fields labeling sssee 13 Port Settigs i ont ori emt t te ee tenni 100 Key fasi LE uds iun ML 99 POMS iecore cien rita hend uir trema ite 100 SATUS T NEN ES 13 Precaultions crei a FRE HE a ge Ix Leges 3 Keys Programmable keys ssseeen 14 eng P R 15 Dialling eee 13 Programmabl
72. e may appear as a secondary line on the OpenStage 5 SIP phone The line type may be Private line type Shared line type i e secondary line on other SIP endpoints Primary line Any line on the Keyset that is a shared appearance of a Primary line on another Keyset but not the Primary line for the Keyset A line that is not characteristic of any specific SIP endpoint i e Phantom line not a primary line on any SIP endpoint May be shared or pri vate Secondary line A gt Page 41 53 gt Page 70 Line key operation mode Select the according line key operation mode Hold If a call is initiated to a phone and speech path is established and then the primary or sec ondary line function key is pressed then the other phone is set to hold and the phone now is in idle state with two options To press the primary or secondary line key that now flashing and reconnect to the other phone or To initiate another call Release If a call is initiated to a phone and speech path is established and then the primary or secondary line function key is pressed then the other phone is cleared Al gt Page 42 F3 gt Page 68 Line monitor The status monitor remains active even during the normal operation of the OpenStage 5 SIP However it does not affect the operation of the function keys Line Monitor information OpenStage 5 SIP LAN Port 1 LAN 100 Mb s Full duplex LED 5 6 7 100 Mb s Ha
73. e should not be confused with the dial plan of the server Phone docu mentation uses the term number plan for the server s dial plan The phone s terminology usage is supported by standards RFC The purpose of the dial plan is to reduce or remove the post dialling delay caused by other meth ods of determining end of dialling either use of a timer or pressing of a done key Both alterna tives are supported by the phone the timer defaults to 6 seconds and the done key is either the tick or lifting the handset The dial plan makes a critical assumption about the number plan it is possible to determine the total number of digits required from the leading digits It does support a variable range of lengths for a given set of initial digits but in this scenario it is not as efficient and brings some restrictions It is possible to create a SIP number plan which prevents effective use of the dial plan If it is de sired to use a dial plan it has to be considered when the number plan is created Dial plans are supported by other manufacturer s SIP phones A well planned dial plan can significantly impact on a user s perception of the system Absence of a dial plan gives the appearance of a slow system A bad dial plan gives a view more akin to sometimes it s very slow the user sees both fast and slow responses The Make up of a Dial Plan Currently a dial plan is restricted to 48 entries This figure was believed to be eno
74. e user types a URI not just a name or number then the entered domain will be used not the Default OBP domain thus providing an override mechanism If this flag is Off but a Default OBP domain is configured it will be ignored If DHCP delivers the address of an SIP Server according to draft ietf sip dhcp 06 txt the server must be treated as an Outbound Proxy server In this case if the Outbound Proxy Flag was Off but the server address was delivered by DHCP this flag would be automatically enabled and both the flag setting and the Server Registrar address would be read only Configuration examples and their behaviors Server Registrar gt Page 114 is in dom1 com and resolves to an IP address w x y Z dom2 com resolves to a b c d OBP OBP Message option domain routed to User input p kelly dom2 com Enabled Not set p kelly dom2 com p kelly dom2 com Disabled Not set p kelly dom2 com p kelly Disabled Not set p kelly dom1 com p kelly Enabled dom2 com p kelly dom2 com p kelly dom1 com Disabled dom2 com p kelly dom1 com f you use an Outbound Proxy you have also to configure the domain name of the Outbound Proxy server gt Page 89 The default setting for the Outbound Proxy flag is Off Al gt Page 38 EI gt Page 63 Part Number Shows the version of the Hardware revision of the OpenStage 5 SIP Al gt Page 51 gt Page 61 108 Alphabetical Reference Payload securit
75. ecause of a congested network they will still arrive at the Reg istrar before the registration expires If the Registration Timer is set to O the phone will not put an Expires header into the REGISTER messages that it sends i e it will not make any suggestion to the Registrar of how long it would like the registration to remain valid Examples Example 1 Configured Registration timer value 3600 secondes Server doesn t set a timer value 1st registration procedure Telephone calculates expires header 3600 20 4320 seconds Telephone gt Server REGISTER Expires Tag 4320 seconds Server gt Telephone 2000K Expires Tag 4320 seconds Re registration 3600 seconds after the first successful registration at the server Example 2 Configured Registration timer value 3600 seconds Server sets the timer to 120 seconds 1st registration procedure Telephone calculates expires header 3600 20 4320 seconds Telephone Server REGISTER Expires Tag 4320 seconds Server gt Telephone 2000K Expires Tag 120 seconds Re registration Telephone calculates expires header 120 20 96 seconds Re registration 96 seconds after the first successful registration at the server Value range table Permitted values numeric Range 0 10 4320 seconds Default value 3600 seconds Al gt Page 38 F3 gt Page 63 Reservation Timer Determines the timeframe for which
76. eeseeeesereesess 30 Terminating line preference sess 122 Testing Connections per line monitor 102 KGyS tee etlenp datum bn tem desi d aay 99 LEDE 1 2 d ped diee dad 101 Perform tests ce endexe enis 51 Lg ST 108 RAM ies d tine a d eder daa ds 111 nis 113
77. egal 2 S 5 Extensions 3 S 4 Extensions 4 S 4 Extensions 5 S 4 Extensions 6 S 2 Communication Group 7 S 6 5 Private Network Trunks 900 S 10 18 5 International 901 S 11 12 3 National 902 S 12 3 National 903 S 11 12 3 National 9118 S 7 Directory Enquiries 99 S 8 Public Local Area 98 S 8 Public Local Area 9x S 2 Illegal S 6 5 Feature Codes S 6 5 Feature Codes The dial plan should begin with a line of up to 14 characters providing a unique identification of the Dial Plan What it can t do You can t have different entries which are used when the phone is locked All entries are pro cessed all of the time Lock bypass is an option added to the basic functionality You cannot bar the sending of digit strings except when the phone is locked There is no mechanism to activate and deactivate individual entries although it is possible to turn the whole plan off A gt Page 42 3 gt Page 61 and gt Page 72 Dial string User Manual gt Page 70 Discarded in outbound packets Displays the number of discard messages according to MIB The used MIB objects are MIB Objects Explanation iflnDiscards Discarded ingoing packets ifOutDiscards Discarded outgoing packets A gt Page 49 Fi gt Page 66 94 Alphabetical Reference Domain Name s the name of the local domain the phone belongs to Value range table Permitted values alphanumeric Length max 92 digits
78. elect a function Function Key assignments gt Page 44 Assign a function to xx Loudspeaker m Web Interface Function key Line key Deactivated Function key Selected dialing Function Key assignments Page 44 Selected dialing 123456654321 Function key Repertory dial A Function Key assignments gt Page 44 Assign a function to key 1 Web Interface 71 Function key Feature Toggle A Function Key assignments gt Page 44 Assign a function to key 6 Feature toggle y 1234567890 Make line busy Stop hunt Apply Function key DSS Precondition A line key has to be configured before Function Key assignments Page 44 Assign a function to key 6 Web Interface Dial plan Dial Plan Configuration and Status gt Page 42 Dial Plan Display file Display memory Feature Access OpenStage 5 SIP example Feature access Auto reconnect CTI S Callback no reply Call deflect Call display by name Call display by number Call duration GPU New Call Beep DSM Speed dial DSM Voice recognition Feature Access gt Page 44 Web Interface 73 Configuration Management Configuration management settings Al Specify configuration update file gt Page 48 172 29 128 156 0 0 0 0 HH OS5 conf 74 Web Interface Non user assisted tests Non user assisted diagno
79. enStage 5 SIP The Chapter explains alphabetically sorted terms that for instance you will encounter in the menus Used symbols A Shows administration tasks Fg Shows the related Web Interface surfaces y Refers to the User Manual This is followed by the Chapter Abbreviations and Specialized Terms Description of Functions Action on submit Select the download type No transfer Upload configuration Download configuration Download application Download NETBOOT A gt Page 46 f gt Page 65 Administrator password Use this function to change the password that is necessary for accessing the administrator area Value range table Permitted values numeric Length min Is predefined by DLS Length max Is predefined by DLS up to 24 digits Default value 123456 A gt Page 53 f gt Page 78 Alphabetical Reference 83 Alert indication Use this function to specify different ring tones for distinctive alert info URLs identifiers Melody Tone Duration Identifier Value range table Permitted values alphanumeric Length max 50 digits Example Strings bellcore dr1 bellcore dr2 see system documentation for bellcore dr3 bellcore dr4 identifier string alert group alert external alert internal alert visual alert emergency alert autoanswer alert priority alert acd alert community 1 alert community 2 alert communi
80. enu 25 Viewing Software download sssssssssss 44 46 ne Diagnostic tests sssssse 51 Specialized terms sssssssssssssssss 125 Symbols in the manual 8 Phone information sssssssssees 51 E VLAN Discovery method sss 124 T Manual VLAN identifier sssssssssse 105 Overview oo eecececcccecccccceceeeeeeecesaeaeaeaussaeeasasaeeseseness 26 Telephone Voicemail number 2 ccceececeeeeeeeeeeeeteeneeaees 124 Call features iii cr teretes 17 Capabilities esses 16 Iristallatlon cete eene 10 W Label m 9 LOCATON P 4 wwen pages Maintenance iee tico aaa ia 4 NODES Io CX gece MEME a Administration interface ssssssssssss 28 Protocols ee i reet 16 Connection establishing 59 Registration ice etes aeds 20 eere Pic yc Uy cem Restart 2 deduce stated toti eds 52 Special configurations sssesssssss 133 Terminal IP address rt eo eret E HIR eR an 121 MASK sce T 121 Restart eec evene ier dtd oen ends 52 Terminal details Name lee i qe oo drea descende 121 Nufnbert deco dde eb dria abl das 121 Register by terminal name sssss 111 Terminal IP Address sssssssssssssss 30 Terminal Mask cccccccccececceeeeeeeesesaeaese
81. er Backup Server is entered and activated Backup IP address or DNS name IP address is entered e g 192 168 1 gt Page 119 gt Page 80 Backup registration The Checkbox for the Backup registration feature is marked gt Page 80 and the feature is activated in the telephone menu gt Page 119 The telephone registers both at the server OpenScape Voice and the backup server In case the OpenScape Voice fails or is not available Both outbound and inbound calls are possible via the backup server Extended Administration 57 SIP Security Configuration Overview Aim secure voice transmission The aim of SIP security configuration is secure voice transmission between telephones Encrypted voice transmission Prerequisite secure signal transmission Secure signalling must first be provided for in order to facilitate secure voice transmission Additional security server authentication The server OpenScape Voice must authenticate with the telephone Server certificate du uM Server certificate Additional feature continuous connection verification This mechanism checks regularly whether the TLS connection is still alive even if the phone is not in use If it determines that the default server is not reachable the phone can switch to a redundant server if e g DNS SRV is used This reduces longer waiting times for the user by avoiding that the phone would have to register with the redunda
82. er IP address Primary line W gt Page 104 Line key configuration dialog Primary line mark as primary if required Ring wW gt Page 104 Line key configuration dialog Ring enable disable Hunt ranking W gt Page 103 Line key configuration dialog Hunt ranking select order User ID W gt Page 104 Line key configuration dialog User ID enter ID Password W gt Page 104 Line key configuration dialog Password enter password Shared type W gt Page 103 Line key configuration dialog Shared type select type 42 Extended Administration Configure Multiline Operation Parameter Registration LEDs W gt Page 111 Web Interface path Menu gt Page 60 e Multiline operation Registration LEDs activate deactivate checkbox Rollover type W gt Page 113 M Multiline operation Rollover type select rollover type Rollover Volume W gt Page 113 e Multiline operation Rollover volume select rollover volume Originating line preference W gt Page 106 UP Multiline operation Originating line preference select line preference Terminating line preference W gt Page 122 oe Multiline operation Terminating line preference select line preference Line action mode W gt Page 102 TM Multiline operation Line key operation mode select operation mode Show focus 27 gt Page 114 E us
83. er can only work with one of the two standards DiffServ or IP precedence for example an older router that only works with IP precedence than the router can translate the ToS field accordingly This can be set for each PSTN peer or LAN interface Ethernet IP header Header DiffServ IP Prece IEEE802 1p dence Decimal Drop level 6 Bit 8 Priority hig med low Bit h A gt Page 36 and gt Page 27 F3 gt Page 64 Alphabetical Reference 111 RAM test Use this function to test the gt RAM memory of your OpenStage 5 SIP The results are displayed after the test Al gt Page 51 3 gt Page 74 Register by terminal name If set to On the phone will send REGISTER messages which contain the contents of the Ter minal Name field gt Page 121 If set to Off the phone will send REGISTER messages which contain the contents of the Terminal Number field Page 121 Al gt Page 37 F3 gt Page 63 Registration backoff timer After the timespan specified here the phone will register anew if the previous registration attempt has failed This timer will be used if a transport connection can be established but the SIP Server is not yet available This should work for UDP TCP and TLS connections This timer will replace the current fixed 1 minute timer which will re send the REGISTER when no response has been receive
84. er or not beeping is heard via the current audio path when an alert ing call is auto answered Mark this option enabled disabled Al gt Page 39 F3 gt Page 64 SIP Beep on auto reconnect This setting controls whether or not beeping is heard via the current audio path when a toggle or alternate between the active call and held call occurs automatically Mark this option ena bled disabled Al gt Page 39 F3 gt Page 64 SIP password Value range table Permitted values alphanumeric Length min 6 digits Length max 24 digits Al gt Page 38 F3 gt Page 63 SIP realm This field displays the realm that the phone is registered in Authentication Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 38 F3 gt Page 63 116 Alphabetical Reference SIP routing When Server is selected the phone will send REGISTER messages to the Registrar server whose address is in the SIP Registrar field If Gateway is selected the gateway address of e g a hardware box is used to phone into the public network direct conversion of SIP to TDM When The phone will not send REGISTER messages Routing type Server If a SIP proxy server is used Gateway If a gateway is used Al gt Page 38 F3 gt Page 63 SIP server type Select the according server type OS Voice Default Other Al gt Page 38 F3 gt Page 63 SIP server vali
85. erately short to explain specific principles A SIP server can be used in one of three ways The nature of the numbers used will vary according to the type of use a PABX replacement a complete private network or a public exchange A PABX replacement in the UK might have only a few entries 2 S 4 B internal numbers can be dialled when locked 3 S 4 B internal numbers can be dialled when locked 9 8 4 20 3 external numbers Notice that minimum lengths need not be specified at input time however the maximum length will be substituted internally compare the results of viewing memory and file on the web page with any of these examples A few refinements can be added according to need There could be an internal emergency number 3333 This has to be placed first It does not conflict with the use of 3 as a first digit Once the user deviates from a sequence of 3 s the search will find the other entry This entry is only required if it is desired to cancel forwarding and DND to allow calls to be returned to the caller 3333 S 4 E internal emergencies Refinements could be made to the public network entries as having critical timing over such a range of lengths might be hard on a caller Also only certain lengths actually exist Nottingham s local numbers are typical of large UK cities smaller cities would have 7 digits while London needs 9 Some of these entries are order sensitive 9999 5 4 E tr
86. eries received from an SNMP manager gt Page 66 SNMP Trap IP address or DNS name f an gt SNMP server exists in the network enter the gt IP address or host name of this server also called Management Center Value range table Permitted values numeric with DNS also alphanumeric Length max 15 digits incl dots with DNS also 92 digits Al gt Page 49 F3 gt Page 66 SNTP server address or DNS name f an 2 SNTP server exists in the network enter the IP address or host name of this server Value range table Permitted values numeric with DNS also alphanumeric Length max 15 digits incl dots with DNS also 92 digits A gt Page 40 3 gt Page 65 Alphabetical Reference 119 Survivability Backup Address IP address of the backup server Has to be entered manually as it cannot be retrieved via DHCP Value range table Permitted values numeric Length max 15 digits incl dots Al gt Page 54 F3 gt Page 80 Survivability Backup Port The Backup port option displays the port number used for SIP communication with the backup proxy server port Al gt Page 54 F3 gt Page 80 Survivability Backup Registration The Backup registration option indicates whether or not the phone treats the Backup proxy serv eras a SIP Registrar If the setting is On the phone tries to register its SIP address with the Back up proxy server Al gt Page 54 F3 gt Page 80
87. es Act as UAS and UAC The OpenStage 5 SIP can initiate SIP requests and re spond to requests Gateways Provide call control Gateways provide many services the most common being translation between SIP conferencing endpoints of transmission format communications proce dures and codecs Other functions include call setup and clearing on both the LAN side and the switched circuit network side SIP Servers SIP servers include the following Proxy servers Receive SIP requests from a client and forward them to the next SIP server in the network Proxy servers can provide functions such as authentication authorization network access control routing reliable request retransmission and security Registrar servers Process requests from UACs for registration of their current location Reg istrar servers are often colocated with redirect or proxy servers 20 Technical Overview Additional Components DHCP server Distributes IP data and further information in a network automatically list of distributed informa tion gt Page 89 SNTP server Provides time date daylight saving and timezone information Messaging server For recording and reading messages SNMP server Logging and maintenance of network components FTP server For up and download of files from and to the phone These include configuration files and music files PC with internet browser Enables the administration of the phone by using
88. ether this parameter is used depends on the configuration of the registrar server Value range table Permitted values alphanumeric Length max 92 digits A gt Page 37 1 gt Page 63 Terminal number The phone will send REGISTER messages containing the contents of this parameter but only when the Register by Name field gt Page 111 is set to Off and the SIP Routing field 2 Page 116 is set to Server Whether this parameter is used depends on the configuration of the registrar server Value range table Permitted values numeric Length min 1 digit Length max 20 digits Al gt Page 37 F3 gt Page 63 122 Alphabetical Reference Terminating line preference Terminating Line selection provides connection of the phone to one of the lines appearing on it on an automatic basis when calls are alerting or ringing audible and the user goes off hook There are five terminating options A Keyset is assigned one of the following preferences Ringing Line Preference A line in the ringing state is selected for terminating calls In the case of multiple lines lines shall be selected on a first in first out basis Ringing Line Preference with prime line preferred Same as Ringing Line Preference but if the prime line is ringing at any time it is signaled and selected before calls on secondary lines Incoming Line Preference In the case of multiple lines alerting or ringing on a device
89. ettings between different keysets on the same shared line cannot occur Mark as primary if correct Al gt Page 41 53 gt Page 70 Line Realm This field displays the realm that the phone is registered in Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 41 1 gt Page 70 Line Ring If the Keyset is not currently in use and the alerting line is allowed to ring the alerting line will get the focus until the call is no longer alerting The audible ringing will be the standard ring as used on a non Keyset phone Mark ringer on off A gt Page 41 3 gt Page 70 Line user ID Enter the according SIP User ID Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 41 gt Page 70 Alphabetical Reference 105 MAC address Displays the world wide unique MAC address of your OpenStage 5 SIP The MAC address is also visible on the label at the bottom of the OpenStage 5 SIP Page 9 Al gt Page 51 gt Page 61 Management Center Port Allows the user to specify the network management port to receive gt SNMP traps sent by the phone s SNMP agent gt Page 66 Manual VLAN identifier Can be defined only if the gt Layer 2 support is activated gt Quality of Service QoS This value describes the association with a certain VLAN if a gt VLAN is used Value range table Permitted values numeric Range 0
90. ey key Receiving a call Reduce the volume of the ringer tone Increase the volume of the ringer tone Open listening Reduce the volume of the loudspeaker Increase the volume of the loud speaker Using the handset of the tele phone Reduce the volume of the handset loudspeaker Increase the volume of the hand set loudspeaker Permanent audio settings Adjust loudspeaker volume ringer volume and handset volume confirmed by key Confirm Restart and factory setting Starts these functions 16 Phone Features Phone Features Protocol support The OpenStage 5 SIP supports the following protocols gt SIP RFC 3261 compliance gt SDP gt TCP gt UDP gt FTP gt SNMP gt SNTP 2 HTTP gt RTP gt RTCP gt DNS gt DHCP gt EAP 802 1X Quality of service in accordance with DiffServe and IEEE 802 1p q Capabilities The OpenStage 5 SIP supports the following capabilities Network Power over LAN DHCP for automatic IP address assignment or static IP configuration SNTP for automatic time synchronization Support for VLANs Support for configurable Layer 2 and 3 Quality of Service Configuration Country definition allowing flexible tone generation Feature enable disable User and administrator levels password protected Upload and download of configuration files INI file format Management Deplo
91. g free ware on the internet Please read the documentation for the FTP software for details of how to install and configure the FTP server Common FTP Server Access Configuration The FTP client on the phone will open a session and therefore requires Account name Username Password Path optional Please note that Account name and Username might be the same on the FTP server used In this case use the name for setting both parameters The parameter path allows you to specify a directory path on the FTP server where the files you want to up or download are located This path is relative to the path set for the user on the FTP server The combination of both settings will make up the full path Example If the user s path on the FTP server is C temp and the path set on the phone is osbMiles the directory where you will need to put the files is C temp os5 files P tameter Web Interface path Menu gt Page 60 Download Server Address E W gt Page 94 File transfer Download server IP address or DNS name enter IP address FTP Account Name W gt Page 96 File transfer m FTP account name enter account name FTP Username y W gt Page 97 File transfer FTP username enter username FTP Password W gt Page 97 File transfer E New Confirm FTP password enter re enter password FTP Path P W gt Page 97 File transfer FTP path enter path FTP Passive Mod
92. he WBM of the phone in the browser DHCP has to be activated to enable this feature Scenarios for Terminal Hostname handling In the default state the option Use E164 as hostname is activated and the default host name set If the E164 entry is input or modified this value is transferred to the Terminal Hostname field The terminal host name can be modified by the administrator even if the option Use E164 as hostname is activated This modification is possible both via the telephone menu and via the web interface Modification for a mobile user is only possible via DLS if the option Use E164 as hostname is deactivated Once a value has been entered to the Terminal Hostname field it is no longer possible to set this field to ZERO not even via DLS If a terminal host name is to be modified via DLS the option Use E164 as hostname must be deactivated for this purpose If the option Use E164 as hostname is activated for a mobile user the E164 number of the mo bile user is entered to the Terminal Hostname field when the data is downloaded If the option Use E164 as hostname is deactivated for a mobile user the value in the Terminal Hostname field is not modified when the data is downloaded This can be useful when a device name is assigned to the IP address in the DNS for example If the option Use E164 as hostname was deactivated for a basic user the value stored in the Terminal Hostname field is used when the
93. he ISDN network PING Abbreviation for Packet Internet Groper A program to test whether a connection can be made to a defined gt IP target Data is sent to the target and returned from there during the test The result of the test displays the success failure of the transmission and possible additional information such as the transmission time PoL PoE Abbreviation for Power over LAN Port Ports are used in gt IP networks to permit several communication connections simultaneously Dif ferent services often have different port numbers QoS Abbreviation f r Quality of Service Additional information see Page 27 RTCP Abbreviation for Realtime Transport Control Protocol RTP Abbreviation for Realtime Transport Protocol RAM Abbreviation for Random Access Memory Memory with read write access ROM Abbreviation for Read Only Memory Memory with read only access SDP Abbreviation for Session Description Protocol SIP Abbreviation for Session Initiation Protocol Protocol standard for initialising calls in gt IP networks Additional information see gt Page 18 128 Alphabetical Reference SNMP Abbreviation for Simple Network Management Protocol The protocol is used for communication with servers that takeover network management func tions This includes for example protocolling errors that occur in network components SN MPTrap Additional information see g
94. he non tagged LAN In these circumstances network routing will probably not be correct the SIP server may or may not be reached The default setting for the phone is to try and perform VLAN discovery DLS The Deployment Service DLS is a OpenScape Management application for administering work points optiPoint OpenStage telephones and optiClient installations in OpenScape Voice net works It has a Java supported web based unser interface which runs on an internet browser Amongst the most important features are security e g PSS generation and distribution within an gt SRTP security domain software deployment plug amp play support as well as error and activity logging Imp For detailed informationen about DLS refer to the OpenScape Voice Deployment Service Ad ministration Manual Impe The Authentication is done via digital Certificates For detailed informationen refer to the IEEE 802 1x Configuration Management Administration Manual Quality of Service QoS Modern networks can be used to provide various Qualities of Service to network endpoints based upon the importance of the endpoint and its generated traffic Quality of Service is a term used to describe this catagorisation of network traffic in networks based on the importance of the data and the treatment of that prioritized traffic A typical example of use of QoS in a network is that of an IP Phone Telephone Voice traffic is more important that for HTT
95. he second ary server is optional The purpose of the secondary server is to allow a backup DNS server to be used in the system environment to increase system availability and reliability The phone can contact several types of server for different reasons SNTP SNMP SIP server etc If a server has been configured by name and not IP address the phone will provide a DNS lookup to resolve the name to an IP address when the server needs to be contacted To optimize network traffic performance the phone caches the result of the normal A and AAAA record lookups and will not re issue a request to the DNS server to resolve that address again until the Time To Live value from the previous lookup has expired If a secondary DNS server has been configured and the primary fails to respond to a request that request will be re issued to the secondary DNS server DNS SRV The phone supports the use of DNS SRV record lookups to allow SIP servers to be located This mechanism is described in detail in RFC 3263 Locating SIP Servers If the location to which a SIP message is to be sent is defined as a name as opposed to an IP address a DNS SRV lookup will be performed An example query being Sip tcp example com This indicates a query for a SIP server supporting the TCP transport protocol The transport used in this query is determined by the SIP transport menu setting Page 117 The DNS server may return an IP address for the requested SIP server or may
96. ils Paranieter Web Interface path Menu gt Page 60 Terminal Number W gt Page 121 SIP environment ii Phone number enter terminal number Terminal Name e W gt Page 121 SIP environment Phone name enter terminal name Register by Name E W gt Page 111 SIP environment l Register by name activate deactivate checkbox 38 Extended Administration SIP Specific Configuration Parameter SIP Routing W gt Page 116 Web Interface path Menu Page 60 SIP environment SIP routing select routing Outbound Proxy W gt Page 107 SIP environment Outbound Proxy activate deactivate checkbox Default OBP Domain Name W gt Page 89 SIP environment Default domain name enter domain name SIP Server Address W gt Page 114 e SIP environment Server IP address or DNS name enter IP address SIP Registrar Address W gt Page 114 e SIP environment Registrar IP address or DNS name enter IP address SIP Gateway Address W gt Page 114 e SIP environment Gateway IP address or DNS name enter IP address SIP Phone Port W gt Page 114 SIP environment SIP Port RTP Base Port enter Port addresses SIP Transport W gt Page 117 SIP environment SIP transport select transport SIP server type W gt Page 116 SIP environment SIP server type select type SIP Realm W gt Page 115 SIP e
97. imminent restart Other configuration options that the phone attempts to retrieve from the DHCP server include Default Route Routers option 3 IP Routing Route 1 amp 2 Static Routes option 33 SNTP IP Address NTP Server option 42 Timezone offset Time Server Offset option 2 Primary Secondary DNS IP Addresses DNS Server option 6 DNS Domain Name DNS Domain option 15 SIP Addresses SIP Server amp Registrar SIP Server option 120 Vendor Unique option 43 Page 23 These parameters are not essential to basic network configuration the operation of the phone and if not obtained will not cause a reboot The phone assumes these parameters are not provided by DHCP until they are returned from the DHCP server If these parameters are returned from the DHCP server they are used and not editable in the various phone menus If these parameters can not be obtained from the DHCP server the manually configured settings for these options are used pe SIP Server option 120 Because the phone only reads the first name IP address supplied in option 120 the maxi mum length of the contents has been limited to 50 octets Please be aware of this when you are using it VLAN discovery per DHCP An additional use for DHCP in the phone is the gt VLAN discovery per DHCP feature This allows the phone to discover its VLAN from a DHCP server in the untagged LAN After discovering its VLAN the phone starts its standard DHCP process within that d
98. inal BOOTP protocol only allowed for the automatic configuration of IP related parameters and for the detection of a server to boot and endpoint from DHCP is a more generic in that it allows for the request of a set configuration op tions and these options are not constrained to the basic IP related parameters Activate this option if the required IP data of the telephone should be assigned dynamically by a gt DHCP server If no DHCP server is available please deactivate this option In this case the data correspond ing to the gt Terminal IP address gt Terminal mask and gt Default Route must be defined man ually List of information obtained by DHCP 90 Alphabetical Reference Basic informations Terminal IP Address Terminal Mask Network Mask Optional informations Default Route Routers option 3 IP Routing Route 1 amp 2 Static Routes option 33 SNTP IP Address NTP Server option 42 Timezone offset Time Server Offset option 2 Primary Secondary IP Addresses DNS Server option 6 DNS Domain Name DNS Domain option 15 SIP Addresses SIP Server amp Registrar SIP Server option 120 Vendor Unique option 43 Changes made at the DHCP Server are not automatically accepted by the phone upon DHCP lease renewal Instead the phone requires a reboot before the DHCP supplied parameters will be applied to the phone Default value On Al gt Page 35 1 gt Page 62 Dial Plan The dial plan of the phon
99. infrastructure A good example is of the data and voice networks being partitioned into data and voice VLANs This isolates the two networks and helps shield the endpoints within the voice network from disturbances in the data network and vice versa VLAN is a layer 2 Physical Layer protocol In the case of Ethernet the physical header is extend ed allowing endpoints to be not only be addressed via MAC address but also VLAN ID gt Page 105 Ethernet VLANs support the partitioning of a physical LAN into up to 4095 virtual LANs To implement a voice network based on VLANs requires the network infrastructure the switch fab ric to support VLANs at layer 2 Dependant on the overall architecture it may or may not be nec essary for the endpoint phone to support layer 2 VLAN The ports of the network switches in the switch fabric can be logically grouped as ports belonging to particular VLAN The switch only forwards traffic to a particular port if that port is a member of the VLAN that the traffic is allocated to In this way an endpoint connected to a particular port on the switch is automatically a member of that VLAN without being a VLAN aware device the switch ensures the endpoint only receives traffic for that VLAN and ensures traffic from the endpoint is only forwarded to ports that are configured to be in the same VLAN This is known as port based VLAN in the switch world VLAN support The phone can be configured as a gt VLAN aware e
100. into 2 per Hop Behavior groups 1 Expedited Forwarded EF referred to RFC 2598 Expedited forwarded is used for voice RTP streams by default High priority traffic to be handeled at the arrival rate DSCP value 10 1 1 1 0 DSCP Diffserv Codepoint Effectively creates a special low latency path in the network 2 Assured Forwarding AF referred to RFC 2597 Assured forwarding is used for signaling messages by default AF31 and less stringent than EF in a multiple dropping system The AF values are containing two digits X and Y AFXY where X is describing the priority class and Y the drop level Four classes X are reserved for AFXY AF1Y High Priority AF2Y AF3Y and AF4Y Low Priority Three drop levels Y are reserved for AFXY AFX1 low drop level AFX2 and AFX3 High drop level In the case of low drop level packets are buffered over an extended period in the case of high drop level packets are promptly rejected if they cannot be forwarded 110 Alphabetical Reference Standard IPV4 Three MSB Called IP 4 Precedence Layer 3 IP Precedence old recommendation replaced by Diffserv IP Precedence is classifying traffic flows into 8 different precedence levels These 3 Bits are the same as the priority Bits from Diffserv Value 5 BIN 101 is used for voice by default And value 3 BIN 011 is used for signaling These values are already configured by configuring the DiffServ values If a routing partn
101. iscovered VLAN to configure itself from the DHCP within that VLAN Technical Overview 23 DHCP Support Explanation of Option 43 As no DHCP option exists for the exchange of VLAN information over DHCP the Vendor Specific Information option 43 shall be used to encapsulate VLAN and download configuration The fol lowing diagram illustrates the format of the Vendor Specific Option Byte 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 De E Option 1 data Siemens identifier ge Opten2 data gt 2 2 2 2 scrip 290 90 290 to be allocated 5 fo fo o v 9 tion O S 9 S 9 o VID v a 8 e e 9 O ju O a O 4 a n a n n O Value 431201 10 S l i e mie n si0 0 0 2 4 0 0 zx CO O 0 0 0 0 255 oo gt gt Byte 1 contains the tag 43 option 1 data contains the Siemens identifier and the VLAN ID is contained in option 2 Five Padding Bytes starting in Byte 20 and the terminate option 255 in byte 25 complete the option frame If you have to specify a configuration download server for configuration update see gt Page 48 so you have to add the values of the server You can place the new option before or after the VLAN information but the five Padding Bytes Value 0 and the termin
102. ity Pre ferred G729 G722 G722 G722 G722 High G722 G722 G722 G722 Quality Pre ferred G723 G722 G722 G722 GT722 Low G729 G729 G729 G729 Band Preferred over 729 G729 G729 G729 G729 Low G729 G729 G723 G723 G723 G723 G723 G723 G723 G723 Band Pre ferred over 723 Low G729 G729 G729 G729 G729 g729 G729 G729 G729 G729 Band Only over 729 Low G723 G723 G723 G723 G723 G723 G723 G723 G723 G723 Band Only over 723 G711 G711 G711 G711 G711 Preferred Codec 729 G711 G711 G711 G711 G711 G711 G711 G711 G711 Preferred Codec 723 G711 G711 G711 G711 G729 G729 G729 G729 G729 G729 G729 G729 G729 G729 G729 Preferred Codec 723 Alphabetical Reference G729 G729 G729 G729 G729 G729 G729 G729 Preferred Codec 729 G729 G729 G729 1 Note that the B party uses a codec in the preferred order of the A party and this regarding RFC 3264 section 6 1 Note that the table is read with Phone A calling Phone B The table is not symmetrical so does not describe the situation if B calls A If a call is cleared because the codecs are not compatible Bandwidth calculation Calculation into dependence of Codec RTP packet size Transmission medium 1 LAN Maximum LAN Overhead Ethernet Fu
103. k PSTN or a voice packet gateway if the phone is used in gateway routing mode gt Page 116 Connecting to the Network Ip You have to connect the OpenStage 5 SIP first to the LAN and then to the power supply The OpenStage 5 SIP has one RJ 45 port labelled 10 100 LAN The port supports 10 100 Mbps half or full duplex connections We recommend that you use the port setting Auto gt Page 100 on the port for auto detection of transferring speed and type of connected cable either straight through or crossed Installing the Phone Connectors on the bottom of the telephone Ethernet port for LAN connection optional with PoL Handset connector Connector for a local power supply unit optional 1 Power over LAN If power is supplied over the LAN cable no local power supply is required 11 Installation Starting up the OpenStage 5 SIP Im The OpenStage 5 SIP phone is to connect to a Switch The phone is working also on a Hub but without a guarantee of quality up The Western plugs of all cable connections must audibly snap into place Plug the short end of the handset cable into the handset and the other end into the connector 2 at the bottom of the telephone and feed the cable through the guide channel in the base unit Only if power not supported by LAN N Use only the plug in power supply unit fitting the OpenStage 5 SIP EU S30122 H7722 X UK S30122 H7723 X
104. k at the current status of transferred files Parameter Application download W gt Page 123 Web Interface path Menu gt Page 60 e General Information File Transfer status Configuration download W gt Page 123 P General Information File Transfer status Config upload W gt Page 123 P General Information File Transfer status MoH download W gt Page 123 P General Information File Transfer status System configuration download W gt Page 123 P l General Information File Transfer status Phone configuration download W gt Page 123 General Information File Transfer status Extended Administration 49 Display Application Version If you want to update the OpenStage 5 SIP Page 44 you should find out the current version of the application software gt Page 51 Use SNMP SNMP Server Configuration Parameter SNMP Trap Address W gt Page 118 Web Interface path Menu gt Page 60 e SNMP Settings Management Center Address enter IP address SNMP Password W gt Page 118 SNMP settings New Confirm Query password enter re enter password Management Center Port W gt Page 105 SNMP settings Management Center Port enter port address Send Generic Taps W gt Page 113 SNMP settings Send Generic Taps to Management Center enable disable Send QDC Taps W gt Page 114 e
105. lect Layer 3 Voice value Layer 3 Signalling only if L3On W gt Page 99 e Quality of Service Layer 3 Signalling select Layer 3 Signalling value Layer 2 Voice only if L2On W gt Page 100 e Quality of Service Layer 2 Voice enter Layer 2 Voice value Layer 2 Signalling only if L2On W gt Page 100 e Quality of Service Layer 2 Signalling enter Layer 2 Signalling value Layer 2 Default only if L2On W gt Page 100 e Quality of Service Default enter Default value gt Changing any QoS settings will take effect as soon as the OpenStage 5 SIP is restarted VLAN Settings Parameter Manual VLAN Identifier wW gt Page 105 Web Interface path Menu gt Page 60 e Quality of Service Manual vLAN identifier enter VLAN ID VLAN Discovery Method W gt Page 124 e Quality of Service VLAN discovery method select VLAN discovery Impe Changing the VLAN Discovery Method will take effect as soon as the OpenStage 5 SIP is restarted Extended Administration 37 LAN Port Settings Parameter Web Interface path Menu gt Page 60 LAN Port Setting E W gt Page 100 LAN Port Settings select speed for port 1 LAN Configure System Information To be granted access to a SIP Server some terminal and SIP related information have to be con figured i First of this the SIP server has to be configured Terminal Deta
106. lf duplex LED 5 6 10 Mb s Full duplex LED 5 7 10 Mb s Half duplex LED 5 A gt Page 52 gt Page 74 Alphabetical Reference 103 Line Address of Record Each line will have a unique SIP Address Of Record AoR and will have an unshifted function key and LED line key assigned to it The assignment of key to line is determined by administra tion Value range table Permitted values numeric Length min 1 digit Length max 20 digits A gt Page 41 3 gt Page 70 Line Hunt Ranking The ordered rank is used to search for a line that is suitable for making a call when making outgo ing calls Multiple lines may be given the same rank Lines that are in rank 1 are the first lines to be considered for use Lines of the same rank are considered for use in key number order Select the according rank 1 bis 10 A gt Page 41 5 gt Page 70 Line Shared type Each line on a keyset may be Private only allocated to that Keyset Would be normally be used for calls made to the Keyset user Shared accessible by several keysets A shared line is an Address of Record which appears on multiple Keysets The SIP server is responsible to coordinating basic call control between the Keyset that have an appear ance of the shared line D a Each line is treated as if the Keyset Operation is Line based the server may change call in formation if the line is Device based Only one call m
107. ll Duplex per direction Protocol Bytes RTP 12 UDP 8 IP 20 802 1Q VLAN Tagging 4 MAC incl Preamble FCS 26 Total 70 Bandwith calculation for Ethernet Values in brackets security enabled Packet size Payload ES Used Ethernet Band Voice Codec e width incl Preamble ms Bytes Bytes kBit s G7 10 80 90 150 160 120 0 128 0 G711 20 160 170 230 240 92 0 96 0 G723 7 30 24 34 1014 114 25 1 30 4 G 729 3 10 10 20 80 90 64 0 72 0 G 729 3 20 20 30 90 100 36 0 40 0 G722 10 80 90 150 160 120 0 128 0 G 722 20 160 170 230 240 92 0 96 0 max values 1 Inthe SIP environment it is possible to activate silence supression In this case the bandwidth calculation values are 2 For G 723 it is not possible to activate silence suppression and the only available sample size is 30 ms 3 In the SIP environment silence suppression is always active currently In this case the bandwidth calculation values are max values 4 For G 722 it is not possible to activate silence suppression Alphabetical Reference 87 2 WAN e g ATM WAN Overhead Protocol Bytes RTP 12 UDP 8 IP 20 ATM Overhead 5 Total 45 Bandwith calculation for WAN e g ATM Values in brackets security enabled Packet size Payload WAN Used WAN Bandwidth Voice Codec Packet size k
108. ncoding e Use this function to select one of the two compression encodings that should be used if the com pressed audio mode was selected see Audio mode Page 84 Selectable values G729 and G723 Al gt Page 50 F3 gt Page 66 Config DLS Port If Deployment service is used enter the port address of the server Al gt Page 48 F3 gt Page 73 Configuration filename Specify the name of the file containing the configuration data of the OpenStage 5 SIP The file must exist in a defined directory on the gt FTP server gt Download server IP address or DNS name gt FTP path Value range table Permitted values alphanumeric Length max 92 digits Default value OS5c Al gt Page 46 F3 gt Page 65 or Config Update DLS IP If Deployment service is used enter the IP address of the server A Page 48 3 gt Page 73 Connectivity check To check the TLS connectivity the value in this field has to be greater than 0 The default value is 120 To deactivate the check function you set the value to 0 Any value greater than 0 activates the function After deactiviation activation the phone has to be restarted If you only changed the value greater than O you do not have to restart the phone e 0 7 off 10 3600 normal range 1 9 is equivalent to 10 23600 3600 A gt Page 52 1 gt Page 77 Date Time Manual definition is necessary only if this information is not transmitted automatically
109. nction SIP Port Access base IP port for receiving amp sending SIP messages RTP Base Port Access base IP port for RTP transport Value range table Permitted values numeric with DNS also alphanumeric Length max 15 digits incl dots with DNS also 92 digits Al gt Page 38 F3 gt Page 63 Alphabetical Reference 115 SIP Auto answer This setting controls whether or not alerting calls can obey any auto answer request signalled for the call Automatic answering will only apply to the primary line of a Keyset Mark this option enabled disabled When you dial a number using the CTI application while Auto Answer is activated the tele phone automatically switches to handsfree mode If Auto Answer is deactivated the telephone will ring first and you must then press the loudspeaker key or lift the handset to dial the number and set up the connection to the other station Al gt Page 39 F3 gt Page 64 SIP Auto reconnect This setting controls whether or not a toggle or alternate between the active call and held call can be signalled to the phone and be automatically applied Automatic reconnect will only apply to the primary line of a Keyset Mark this option enabled disabled Use this option for placing a call on hold and for retrieving it again using both a CTI application and the telephone Al gt Page 39 F3 gt Page 64 SIP Beep on auto answer This setting controls wheth
110. ndpoint by enabling VLAN support via the con figuration menus The following VLAN related configurations can be achieved Manually configured L2 VLAN only Manually configured L2 VLAN and QoS Automatically discovered VLAN and manual QoS To configure manual L2 VLAN only the phone must be configured at manual VLAN ID between 1 and 4095 Vlan discovery mode must be set to manual To configure manual L2 VLAN and QoS the phone must be configured as QoS layer 2 on and a manual VLAN id between 1 and 4095 Vlan discovery mode should be set to manual and QoS lay er 2 and 3 values should be configured as described in the QoS section below If you mis configure a phone to an incorrect VLAN the phone will behave as though it is not con figured for and possibly not connected to the network In DHCP mode it will behave as though the DHCP server cannot be found in fixed IP mode no server connections will be possible To automatically discover a VLAN ID using DHCP the phone must be configured as DHCP en abled and VLAN discovery mode set to DHCP If QoS is required this can be turned on and QoS layer 2 and 3 values should be configured as described in the QoS section below Technical Overview 27 The DHCP server must be configured to supply the Vendor Unique Option in the correct Unify VLAN over DHCP format If a phone configured for Vlan discovery by DHCP fails to discover its VLAN it will proceed to con figure itself from the DHCP within t
111. ng the OpenStage 5 SIP Using the OpenStage 5 SIP Speaker Keys for for ring tones telephone settings Handset LEDs Key field Eight freely programmable keys AA Telephone Modes Your administrator can configure the OpenStage 5 SIP for use as A SingleLine phone with one line AMultiLine phone with up to 8 lines in relation with the SIP server Dialling Keypad The dialling keypad of the OpenStage 5 SIP is labeled with digits letters and some special char acters Generally you use the keypad to dial numbers 14 Using the OpenStage 5 SIP Programmable Keys The OpenStage 5 SIP is equipped with function keys which are user programmable in two levels see User Manual These keys come already preassigned in the first level Loudspeaker Voice Message Cancel Release Confirm Blind Transfer oN O OC CQ N Function Key Function Switches the handset microphone on off Switches the loudspeaker on off Indicates and starts voice messages Cancels an active call latest activity Confirms input made Transfers a call without announcement Holds a call explicitly Toggles between first and second key levels 1 2 3 4 5 6 7 8 Using the OpenStage 5 SIP Control Keys The two control keys and C are located on the left side of the dialling keypad Depending on the operating mode you can vary the following settings Operating mode k
112. ngs The following procedure can be invoked in order to reset the OpenStage 5 SIP back to its default factory settings Attention A factory reset deletes all administration data passwords except reset password and user configurations IP and SIP connections will be lost To avoid the necessity of re entering the phone configuration manually after restoring factory settings use the Upload Configuration function Page 46 to save the configuration After factory reset the FTP password is replaced with the string 123abc To perform a factory reset Remove the LAN connection Page 10 D When Power over LAN do not disconnect the LAN Press hard keys 2 8 and 9 simultaneously Press the hard key Enter the standard factory reset password 124816 Terminate by pressing the hard key You also can use the Web Interface to reset factory settings gt Page 78 Change Administrator Password Web Interface Mese oo t pamer o WSD o gt Page 60 Parameter Admin Password W gt Page 82 ome administrator password enter current and new password and confirm Reset User Password Parameter Web Interface path CLE gt Page 60 Reset User Password W gt Page 78 fReset user password enter new password and confirm 54 Extended Administration Clear ALL user data Web Interface path Menu gt Page 60 Parameter Clear ALL user data E W gt Page 87 Clear ALL user da
113. ns in this section are general troubleshooting tips If using a gt DHCP server make sure that it is operating correctly The DHCP server should show an incoming request from the MAC address listed on the product label If you do not see the idle menu after you connect the Ethernet cable make sure the power cord and the Ethernet connection are secure If you do not hear a dial tone make sure that the telephone handset line cord is plugged into the IP Phone port Also make sure that the appropriate Ethernet cable crossover or straight through is used and that all cable connections are secure A fast busy tone indicates that the number you called is not valid or that external circuits are busy Verify the number or try your call again later If you place a call to another IP telephone hear ringback and the called party answers but you cannot hear the speaker s voice Incompatible terminal verify that the OpenStage 5 SIP and the other IP telephone support at least one common audio codec Page 84 In some cases it can be useful if you perform a update of the phone s software Page 44 Fault Finding This section provides guidance of identifying the source of a problem which is affecting the phones ability to function correctly A description of the various faults and possible actions are described below No Registration with Line Keys A line key does not register when digest authentication is activated The reason for this is that re
114. nstructions given may result in injury or in damage to the unit VJ Key information important for the proper use of the unit is marked with this symbol Shows administration tasks on the Web Interface Shows additional information about each parameter in the Alphabetical Reference Shows the related web pages Means that you are in the administration menu and you have already entered the correct admin istrator password Access Web Interface gt Page 60 Means that you are in the diagnostics menu and you have already entered the correct administra tor password Access Web Interface gt Page 60 Means that you are in the setup menu and you have already entered the correct user password if required General Information RIEN CB or Operating the telephone Lift the handset off hook Replace the handset on hook Conduct a call Enter a telephone number or code Increase or reduce the value depending on the current operating mode Changing and viewing the configuration data in the phone is done by entering different reference numbers For description of viewing data values on the LEDs of the entry see gt Page 81 Intended Use The OpenStage phone is a desktop unit designed for voice transmission and for connection to the LAN It can also be used as a workstation device Any other use is regarded as unauthorized Product Identification The identification details of your telephone are given on the
115. nt server first before a call can be set up TLS verified TLS verified 58 Extended Administration Implementation OpenStage 5 SIP telephone settings The following settings must be performed on the telephones for which voice encryption is to be enabled Set payload security to allowed gt Page 77 Set connectivity check interval as required Page 77 0 when deactivated 10 3600 duration of verification in seconds 1 9 10 gt 3600 3600 Set SIP server validation as required Page 77 Set SIP transport to TLS gt Page 63 Set SIP port to 5061 gt Page 63 In the case of web settings references to the corresponding locations are provided Configuration via DLS The security settings can also be configured using the DLS DeploymentService For details please refer to the DLS Administration Manual OpenScape Voice settings See also Test Configuration and Connectivity Solution Manual Web Interface 59 Web Interface Establishing the Connection to the Phone You can display and configure device and network information for the OpenStage 5 SIP through the Web Interface You can access the Web Interface using one of the following web browsers Microsoft Internet Explorer recent version recommended Mozillla Firefox recent version recommended To access the Web Interface perform the following steps Open a web browser and enter the URL of the web page for the phone as follows
116. nvironment SIP realm enter realm name SIP User ID W gt Page 117 SIP environment SIP user ID enter ID SIP Password W gt Page 115 SIP environment New Confirm SIP password enter re enter password SIP Session Timer W gt Page 117 SIP environment SIP session timer enabled activate deactivate checkbox SIP Session Time W gt Page 116 SIP environment SIP session timer value enter time Registration Timer W gt Page 112 A SIP environment Registration timer value enter time Transaction timer W gt Page 122 A SIP environment Transaction Timer enter time Registration backoff timer W gt Page 111 A SIP environment Registration backoff timer enter time Extended Administration 39 D Changing either the Terminal Number or the SIP Routing setting will take effect as soon as the OpenStage 5 SIP is restarted SIP Features Parameter Call handling options Web Interface path Menu gt Page 60 Auto answer w gt Page 115 e SIP features Auto answer mark to be enabled Beep on Auto answer W gt Page 115 e SIP features Beep on Auto answerl mark to be enabled Auto reconnect gt Page 115 e SIP features Auto reconnect mark to be enabled Beep on Auto reconnect W gt Page 115 e SIP features Beep on Auto reconnect mark to be enabled Group pickup Group
117. ocus V Use LED to indicate Remote Forwarding Note set M reservation timer to Reservation timer feo seconds p to disable line reservation Reset Function keys A Function Key assignments gt Page 44 Phone function key assignments Key Normal function Shifted function voe Ex 2 Loudspeaker Est 3 Voice Messages sit 4 Cancel Release Est 5 Confirm Est 6 Blind Transfer Edit 7 Hold Est 8 amp Shit Edit Shit Parameters Function Mute assigned to Key 1 has no parameters Name Value I a Other function keys are Function key Line key gt Page 70 Selected dialing Function key Selected dialing gt Page 70 Reppertory dial Function key Repertory dial gt Page 70 DSS Function key DSS gt Page 71 Keys can be locked by Administrator for the user can t change the contents Web Interface 69 Example Assign a function to xx Do not disturb v Function key Select a function Assign a function to xx Blind Transfer Callback Cancel callbacks Cancel Release Clear definition Confirm Consult Transfer Do not disturb DSS Feature toggle Group pickup Hold Line Loudspeaker Phone lock Repeat dialing Repertory dial Ringer off Selected dialing Shift Function key Select a function See also gt Function key S
118. or more information refer to OpenScape Voice QoS Data Collection V1 0 Interface Description 2 Page 75 Alphabetical Reference 109 QDC Port Enter port address of the Quality Data Collection Server For more information refer to OpenScape Voice QoS Data Collection V1 0 Interface Description 2 Page 75 Quality of Service QoS The QoS technology based on layer 2 and the two QoS technologies Diffserv and TOS IP Prece dence based on layer 3 are allowing the VoIP application to request and receive predictable ser vice levels in terms of data through put capacity bandwidth latency variations jitter and delay Please note that all these technologies are just marking packets which allow the network to clas sify and prioritize the packets accordingly This means that the network decides which QoS con figuration marking will be used and should be set in the endpoints The default values are well known recommendations Layer 2 802 1p QoS on layer 2 is using 3 Bits in the 802 1q p 4 Byte VLAN Tag which has to be added in the Eth ernet header The CoS class of service value can be set from 0 to 7 The value 7 is describing the highest pri ority and is reserved for network management Value 5 is used for voice RTP streams by default Value 3 is used for signaling by default Three Bits Used for CoS Layer 3 Diffserv Diffserv is classifying traffic flows like voice RTP streams or signaling messages
119. oved from the penalty box and it can be tried once again to see if it is back in service Note this mechanism is independent of call setup The first SIP server will not be retried necessar ily when the next call is established only when it is removed from the penalty box SNTP The phone support gt SNTP The SNTP server address can be supplied via DHCP or manually configured gt Page 118 If the SNTP server address is available the server will be queried for the time If a server address is not configured the phone will look for SNTP broadcasts and setup the time accordingly if these are received any manually configured time and date information would be over written SNMP The phone provides gt SNMP which allows network related information to be browsed MIB II sup port Standard SNMP browsers are sufficient for this purpose 26 Technical Overview IP Network Configuration Routing The phone allows a default route to be configured to allow access to Servers on a different subnet to the one in which the phone resides In addition it is possible to configure 2 additional routes Each route consists of a IP address gateway and mask Virtual LAN VLAN VLAN or virtual LAN is a technology that allows network administrators to partition one physical network into a set of virtual networks or broadcast domains Physically partitioning the LAN into separate VLANs allows a network administrator to build a more robust network
120. ow are some specific details relating to SIP registration configuration parameters found on the phone Terminal Number gt Page 121 Terminal Name gt Page 121 Register by Name gt Page 111 SIP Routing gt Page 116 SIP Registrar SIP Addresses gt Page 114 SIP Realm gt Page 115 Registration Timer gt Page 112 21 Technical Overview If a registration attempt should result in a timeout The phone waits a random time before sending another REGISTER message The Reg backoff seconds parameter determines the maximum waiting time 22 Technical Overview IP Network Protocols DHCP The Phone contains a Dynamic Host Configuration Protocol DHCP client that supports automatic configuration of various parameters If DHCP is enabled in the phone the phone will try to obtain the following options that are essential for the configuration of its Ethernet interface automatically from a DHCP server Terminal IP Address Terminal Mask Network Mask When the telephone requests its IP address it sends apart from other information its default host name to the DHCP server The default host name consists of telephone model type MAC address e g OST5D00016325a845 The DHCP server forwards this name to the DNS server together with the IP address assigned If the phone fails to configure its Ethernet interface from a DHCP server it will eventually time out indicating no DHCP server found and
121. pPhone to assign the DLS server address or the name This enables a more specific configuration of the DHCP serv er Impe The Authentication is done via digital Certificates For detailed informationen refer to the IEEE 802 1x Configuration Management Administration Guide 24 Technical Overview DNS The phone uses gt DNS services provided the phones operating system to perform the following Resolve the IP address of servers that have been configured as names DNS A AAAA records Resolve the IP address of the domain part of users called by URL DNS A AAAA records Identify the location of servers and provide for failover and load balancing DNS A AAAA and DNS SRV For DNS services to be used on the phone the following must be configured either manually or provided by DHCP DNS Domain Name gt Page 94 Prim DNS IP addr gt Page 108 Sec DNS IP addr gt Page 113 Impe If a DNS Domain Name and one or both of the DNS IP addresses have been configured then additionally the following host names can be entered alternatively to the corresponding IP addresses SIP addresses Page 114 server gateway registrar Download server IP address or DNS name Page 94 SNMP Trap IP address or DNS name Page 118 SNTP server address or DNS name gt Page 118 Message Waiting IP address gt Page 105 Ping gt Page 108 The Primary DNS server IP address must be configured if DNS is to be used however t
122. rator to view the feature access settings and potentially activate or deactivate individual features on the phone When a feature is deactivated it is no longer available at user level If a feature is activated so it is available at User level its status is shown as On If the feature status is shown as Off the feature is not available at User level and is not displayed as an option in the Configuration menu A gt Page 44 1 gt Page 72 Feature Code For Feature Toggle Enter the code for the required OpenScape Voice feature which you would like to assign to the function key OpenScape Voice supports the following features for example Make line busy the phone number is treated as busy for the hunt group Stop hunt the phone number is removed from the hunt group gt Page 71 Feature toggle User Manual gt Page 71 Forwarding Indication Only for the forwarding function of the OpenScape Voice If this function is activated a blinking line key indicates the forwarding on all phones where a forwarded primary or secondary line is active Al Page 42 1 gt Page 68 FTP account name Refer to the documentation of your FTP server for information about the FTP account Value range table Permitted values alphanumeric Length max 92 digits Al gt Page 45 F3 gt Page 65 FTP passive mode If the passive FTP also passive mode is activated the telephone transmits a PASV command the server opens
123. return a single name or list of names which require further A or AAAA record lookups to determine an IP address The response to a DNS SRV query will also contain information regarding the Time To Live for the information re turned the port address to which requests should be sent and weighting information relating to load balancing of requests Technical Overview 25 DNS SRV and failover Lists of candidate SIP server names are often returned in response to DNS SRV queries to allow failover mechanisms to be implemented which increase overall system availability If the phone sends a request to the first address in the list but fails to receive a response the failover time is configurable default is 32 seconds see also gt Page 122 the address is placed in a penalty box which means that it will not be tried again until a specific time interval has past currently pre set to 10 minutes The request is sent to the next SIP server in the list and the pro cess continues The penalty box mechanism ensures that the responsiveness of the phone is maintained by not continually retrying SIP servers that are failing to respond For example the request to the first SIP server in the list fails a call to another user hosted on the SIP server will result in the user experiencing a 6 second delay before the failover to the secondary occurs All subsequent messages for this call will go to the second SIP server until the first SIP server is rem
124. route gt Page 30 Troubleshooting 131 Subscriber identity not set The subscriber identity number or name is not configured Code 7 D gt Possible solution t gt Enter number or and name gt Page 63 D D No network The telephone cannot find the network Code 8 b gt Possible solution X Check the network cable gt gt D SIP Server address not set The DHCP setting is switched off an a SIP server has not been configured Code 9 gt Possible solution b gt Configure SIP server address gt Page 63 gt y SIP Server not responding The SIP Server Address has been set but the SIP server is not responding Code 10 gt Possible solution D Check SIP realm gt Page 63 D SIP registration error The SIP server proxy has rejected registration of the phone Code 11 gt Possible solution X Check SIP user ID and password Page 63 z z ctw TS A 132 Troubleshooting Common problems Phone Can Not Contact Host Names Ensure that the DNS Domain Name is correctly configured gt Page 94 Ensure that the Prim DNS IP addr is correctly configured gt Page 108 and can be pinged 2 Page 108 Poor Speech Quality Poor speech quality can be the result of an overloaded network Consider the implementing VLANs gt Page 105 and QoS gt Page 109 Ensure that all IP endpoints in your system including SIP Gateways are using
125. s originally developed in the MMUSIC group within the IETF Internet Engineering Task Force it has been published since February 1999 as RFC 2543 The SIP working group is con tinuing to enhance the protocol and published version 2 as RFC 3261 in 2002 SIP Functions Systems which use SIP are able to provide the following The location of the target endpoint SIP supports address resolution name mapping and call redirection The media capabilities of the target endpoint Via Session Description Protocol SDP SIP de termines the lowest level of common services between endpoints Conferences are established using only the media capabilities that can be supported by all endpoints A session between the originating and target endpoint If the call can be completed SIP es tablishes a session between the endpoints SIP also supports mid call changes such as adding another endpoint to the conference and changing media characteristic or codec 19 Technical Overview Components in a SIP system SIP server e proxy S redirect registrar g DHCP server b s SNTP server Messaging server s SNMP server gil FTP server Switch LP OpenStage 5 SIP nm tT PSTN SIP gateway Configuration example with additional components Page 20 SIP Components SIP is a peer to peer protocol The peers in a session are called user agents UAs SIP Clients SIP clients include the following Telephon
126. stic tests gt Page 51 Non User Assisted Tests 182 168 1 151 User assisted tests User assisted diagnostic tests gt Page 52 User Assisted Tests Web Interface 75 QoS Data Collection QoS Data Collection FI L n200 eo 20 is mo am B moo 2o Bo is Web Interface Session data These parameters are used for development only 2 Page 75 EE Submit Select a report to view Aelii Sessondata Web Interface 77 Fault Investigation Available tracing menus Trace Configuration Trace View Simplified trace page FTP Client Exception Data Windview Configuration SIP UDP Trace For information how to trace the OpenStage 5 SIP please refer to the Tracing guide Simplified trace page EasyTrace Settings Timeout minutes 15 File Size bytes 65536 Trace setting template none Y Automatic clear before start o for easytrace Submit Start logging Reset settings For information how to trace the OpenStage 5 SIP please refer to the Tracing guide Security A Security settings gt Page 52 Security Settings WARNING If you enable or disable the fields marked with an asterisk you will have to restart the terminal manually before they take effect Payload security allowed Iv Connectivity check interval T fo seconds 10 to 3600 SIP server valida
127. t Page 25 SNTP Abbreviation for Simple Network Time Protocol The protocol is used between timeservers and terminal devices of a network to synchronize the time of the terminal device Additional information see gt Page 25 SRTP The Secure Real time Transport Protocol is a profile of the Real time Transport Protocol RTP which can provide confidentiality message authentication and replay protection to the RTP traffic and to the control traffic for RTP the Real time Transport Control Protocol RTCP More Informa tion see RFC 3711 Subnet Mask Classifies networks in A B and C networks Each class has a subnet mask that demasks the relevant bits 255 0 0 0 for Class A 255 255 0 0 for Class B and 255 255 255 0 for Class C Ina Class C network for instance there are 254 gt IP addresses Switch Network device that selects a path or circuit for sending data to its next destination A switch may also include a router function TOP Abbreviation for Transmission Control Protocol TLS Abbreviation for Transport Layer Security This protocol ensures privacy between communicating applications UDP Abbreviation for User Datagram Protocol VLAN Abbreviation for Virtual Local Area Network Additional information see Page 26 VoIP Abbreviation for Voice over IP E g voice transmission through IP technology Troubleshooting 129 Troubleshooting General Troubleshooting Tips The suggestio
128. t tries to address the 1st destination again and will again step to the 2nd one if the messages are not answered Al gt Page 38 F3 gt Page 63 Alphabetical Reference 123 Transfer on Ringing If this function is active a consultation can be transferred after you have dialled the third partici pant s number but before the third party has answered the call Al gt Page 39 F3 gt Page 64 Upload Configuration Use this function to save back up an OpenStage 5 SIP configuration on the gt FTP server The following parameters must be set before the upload operation gt Download server IP address or DNS name gt FTP path gt Configuration filename gt FTP account name gt FTP username gt FTP password Al gt Page 46 3 gt Page 65 Upload Download Status Shows the status of the following downloads with the date of the last transfer Application file download Configuration file download Configuration file upload Al gt Page 48 F3 gt Page 61 Use dynamic hostname concept This option is to be considered in combination with the entry in the Terminal Hostname field Fur ther information is provided on Page 120 Al gt Page 35 F3 gt Page 62 Versions Info Displays some telephone versions like Application version SIP stack version SIP signalling version Web content version Netboot version Part number The application version identifies the release level
129. ta confirm with OK Port Control Web Interface path Farameler Menu gt Page 60 Port Control e W gt Page 108 Port control Service Agent Test Interface SNMP Interface Resilience and Survivability To allow for stable operation even in case of network or server failure OpenStage 5 SIP has the capability of switching to a fallback system The switchover is controlled by configurable check and timeout intervals Survivability is achieved in 3 different ways DNS SRV can be used for enhanced survivability either in a scenario with a survivability proxy or in a scenario with multiple primary SIP servers The DNS server provides the phone with a priori tized list of SIP servers via DNS SRV The phone fetches this list periodically from the server de pending on the TTL time to live specified for the DNS SRV records To enable DNS SRV requests from the phone please make the following settings Specifiy the IP address of at least one DNS server that provides the server list via DNS SRV The web interface path is Network and IP Routing Primary Secondary DNS IP address Enable the use of an outbound proxy for routing outbound requests The web interface path is SIP Environment Outbound proxy Set the SIP gateway port to 0 The web interface path is SIP Environment gt SIP details gt SIP gateway Additionally if the SIP server specified in SIP Environment SIP registrar server ad dress is to be configure
130. that may be logically associated with a supported function A function key can support a second function Normal function Shifted function A Page 44 1 gt Page 68 Group pickup URI To be a member of a Call Pickup group for a notification service or the user initiated service the phone must be configured by administration with the URI of the call Pickup group service which is provided by the server Enter the URI of the Call Pickup group lt groupcallpickup lt SIP Server IP e g groupcallpickup 172 16 127 95 or Domain Name Al gt Page 39 F3 gt Page 64 98 Alphabetical Reference Hot line for selected line Mark Checkbox if a hotline is defined for this line 2 Page 70 Hot Warm line default dial string This control allows the administrator to enter a default dial string associated with the phone to be used in connection with Hot Line or Warm Line working Al gt Page 39 F3 gt Page 64 Hot line dial string for selected line This is the hotline target for this line Can only be set by administrator in web interface 2 Page 70 Hot Warm Phone This dropdown allows the user to specify the Hot Line Warm Line operation of the phone There are three options Normal line Action Any Hot Line Warm Line parameters associated with the phone will be ig nored regardless of whether or not the phone is in a keysystem group Warm line If there is a default dial string set against the phone then it will be a
131. tic detection 10 Mbit s full in 10 Mbit networks in full duplex process 10 Mbit s half in 10 Mbit networks with half duplex process 100 Mbit s full in 100 Mbit networks in full duplex process 100 Mbit s half in 100 Mbit networks with half duplex process 1 The data can be transmitted and received simultaneously 2 The data can only be transmitted or received The change will only have effect if you restart the phone Al gt Page 37 F3 gt Page 67 Alphabetical Reference 101 Last Restart e Use this function to view the date and time of the last restart gt Page 52 of the OpenStage 5 SIP Al gt Page 52 LED test Run this test to check the function of the gt LEDs at the OpenStage 5 SIP During the test all LEDs are flashing The Stop button terminates the test A Page 52 3 gt Page 74 Line key Basic terminology Call Appearance The standard OpenStage 5 SIP single line device supports 2 call instances per line When the line is idle there will be no call instances In contrast an analogue wired line only supports a single Call Appearance at a time Multi Call Appearance Future Support of multiple line keys associated with the same Address of Record AoR on a single OpenStage 5 SIP Example AoR of the primary line appears on multiple line keys on a single device Function key A key on the OpenStage 5 SIP phone that may be logically as sociated with
132. ting between local and long distance numbers so it might include 9 3 15 3 feature codes 1 5 3 15 3 feature codes 2 911 S 3 E police X11 C 3 3 3 4 special service numbers X1 5 10 long distance type 1 X0 S 10 long distance type 2 X S local This sequence is order critical except that 911 could be moved higher What it doesn t do well Variable length local numbers involving subsets give a poor performance needing the critical timer to dial the shorter numbers If you have both 2 digit and 4 digit numbers beginning with 2 you need 2 5 2 4 3 Another drawback of this method is that the phone will send 3 digit numbers and these are not valid You could make extensive use of the check function 21 C 2 2 22 C 2 2 23 C 2 2 24 C 2 2 25 C 2 2 26 C 2 2 27 C 2 2 28 C 2 2 29 C 2 2 20 C 2 2 2 5 4 This is cumbersome and would cause the table to be too big if other digits followed this practice or the lengths were 3 and 5 you d need 100 entries to handle the three digit values The dial plan can not be edited on the phone or via the Web Interface The dial plan is a CSV file To load the dial plan to the phone the deployment service DLS has to be used Alphabetical Reference 93 Another example for an US Dialplan below Dialplan us csv IUK Pri Vpp pp 555 S Emergency 999 S Emergency 0 S Operator 1 S Ill
133. tion B Submit Reset settings Restart terminal A Restart the OpenStage 5 SIP gt Page 52 Restart terminal This operation will restart the terminal immediately You will lose your web connection to the terminal Restart 78 Web Interface Reset user password A Reset User Password gt Page 53 Reset User Password This operation will reset the user password to the default value Reset User Password Change admin password A Change Administrator Password gt Page 53 Change administrator password Current password New password Confirm new password Note The password must be numeric and be at least 6 digits long Clear ALL user data A Clear ALL user data gt Page 54 Clear ALL user data This operation will clear out all personal data relating to the user including Personal Directory entries and structure definition ready for the phone to be given to a different user It does not clear out Phone settings such as contrast or touch screen calibration settings nor Administration settings such as network details Restore factory settings A Restore Factory Settings gt Page 53 Restore factory settings Please enter reset password Web Interface Port Control Port Control gt Page 54 Port control Service Service Agent Test Interface SNMP Interface FP
134. to idle mode or calls a specified number if Warm line is active when the handset was lifted or the speaker key was pressed and no number was dialled Al gt Page 39 F3 gt Page 64 Alphabetical Reference 99 Invalid in outbound packets Displays the number of error messages according to MIB The used MIB objects are MIB Objects Explanation iflnErrors Non valid ingoing packets ifOutErrors Non valid outgoing packets Al gt Page 49 Fi gt Page 66 IP routing To have constant access to network subscribers of other domains you can enter a total of two more network destinations An gt IP address of the domain and gateway and a gt Subnet Mask must be entered for each further domain you wish to use Use this function to define the following IP addresses for Route 1 2 IP address IP address IP addess of the selected route Gateway IP address of the gateway for this route Mask Network mask for this route Value range table Permitted values numeric Length max 15 digits incl dots Al gt Page 35 F3 gt Page 62 Key test Test to check the functions of the telephone keys Al gt Page 52 1 gt Page 74 Layer 2 3 The QoS technology based on layer 2 and the two QoS technologies Diffserv and TOS IP Prece dence based on layer 3 are allowing the VoIP application to request and receive predictable ser vice levels in terms of data through put capacity bandwidth
135. ty 3 alert community 4 Alert Type List of options Ring In case of this type the phone rings Silence In case of this type the phone does not ring Ringer Value range table Permitted values numeric Range Melody 1 8 Tone sequence 1 standard melody 2 single shot tone 3 silence Default value Melody 2 Tone sequence 2 To hear the configured melody and tone sequence you have to proceed the according call Duration Value range table Permitted values numeric Range 0 300 seconds Default value 60 seconds Al gt Page 50 f gt Page 67 Append codes The following buttons represent the possibility of the append codes OK Clear Pause y User Manual 5 gt Page 70 84 Alphabetical Reference Application filename Specify the name of the file containing the software of the OpenStage 5 SIP The file must exist in a defined directory on the gt FTP server gt Download server IP address or DNS name gt FTP path Value range table Permitted values alphanumeric Length max 92 digit Default value OS5a Al gt Page 46 F3 gt Page 65 Audio loop test The test activates the microphone and the loudspeaker in the handset You can check these components by speaking and listening Al gt Page 52 gt Page 74 Audio mode Use this function to select the audio transfer codec Audio Mode High Q
136. u n F a Pyg ts Ot d as D A e hea Fm rpc a A ME q LI Ap wv ZU PE E Gd 4 OpenStage 5 SIP OpenScape Voice Administration Manual A31003 S2000 M104 3 76A9 ie A E So fi be EA ft 31 TRI Te u n F uU your enterprise Our Quality and Environmental Management Systems are implemented according to the requirements of the ISO9001 and 18014001 standards and are certified by an external certification company Copyright Unify GmbH amp Co KG 10 2014 Hofmannstr 51 81379 Munich Germany All rights reserved Reference No A31003 S2000 M104 3 76A9 The information provided in this document contains merely general descriptions or characteristics of performance which in case of actual use do not always apply as described or which may change as a result of further development of the products An obligation to provide the respective characteristics shall only exist if expressly agreed in the terms of contract Availability and technical specifications are subject to change without notice Unify OpenScape OpenStage and HiPath are registered trademarks of Unify GmbH amp Co KG All other company brand product and service names are trademarks or registered trademarks of their respective holders unify com Safety Precautions 3 Safety Precautions Important Notes N Do not operate the telephone in environments where there is a danger of explosions valid
137. ualitiy Preferred Uncompressed audio transmission Low Bandwith Preferred Use preferred compressed audio transmission narow band Low Bandwith only Use compressed audio transmission only narow band G711 Preferred Uncompressed audio transmission narow band G729 Preferred Compressed transmission band of about 8 kbit s The value of compression encoding is depending on the selected compression codec see Com pression encoding Page 88 Default value G711 preferred Il Audio codec G 711 If the country code is set to US the audio codec G 711 ulaw is preferred All other country codes causes the audio codec G 711 alaw to be preferred Audio codec G 722 This speech codec offers a wider audio bandwidth resulting in major improvement in the rep resented speech quality Alphabetical Reference 85 Codec Negotiation The following table indicates which codec will be selected depending on which codec is selected on each phone a G711 always option is included because some non Unify phones may include this option Pho High High Low Low Low Low G711 G711 G729 G729 Quali Quali Band Band Band Band Pref Pref Pref Pref Pho ty ty Pre Pre Only Only ered ered ered ered Pre Pre ferred ferred over over over over over over ferred ferred over over 729 723 729 723 723 729 Co Co 729 723 dec dec G729 G723 High G722 G722 G722 G722 Qual
138. ugh while avoid ing excessive searching Each entry consists of a number of fields Leading digits a string to match the dialled digits against Both amp can be included There is also the wildcard X to represent any single digit There is no any sequence value Action s Originally there were two but these have been joined by a third S Send the digits when the maximum digits have been received or if the timer expires after the minimum digits have been received or on receipt of the terminator after the minimum digits C Check for other actions Minimum and maximum must match the length of the string The timer is run at this point only sending will occur on expiry If more digits are received further entries will be checked It is possible to use C in combination with D and have no timer Alphabetical Reference 91 D Give secondary dial tone when the leading digits match This action does not depend on later parameters In theory there is a choice of 9 tones but only digit 1 is currently valid D can be used in combination with C or S Minimum length Automatic sending will not occur until at least this many digits have been di alled Maximum length Automatic sending will occur when this many digits have been dialled Timer A shorter than normal interdigit timer to be used once the minimum number of digits have been dialled or when the check function occurs The timer can be zero
139. utomatically di alled after the delay set with Initial Digit Timer when the user lifts the handset This occurs regardless of whether or not the phone is in a keysystem group Hot line There are two cases to consider depending on whether the phone is part of a keysys tem group 1 If the phone is part of a keysystem group there may be a Hot Line dial string associated with each of the Line keys on the phone If there is a Hot Line dial string associated with the line then it will be automatically dialled immediately when the line is manually selected fthere is not a Hot Line dial string associated with the Line but there is a Default dial string set against the phone then the Default dial string will be automatically dialled immediately when the line is manually selected If there is not a Hot Line dial string associated with the Line and there is not a Default dial string set against the phone then the user will receive dial tone when the line is manually selected 2 Ifthe phone is not part of a keysystem group If there is a Default dial string set against the phone then it will be automatically dialled im mediately when the line is manually selected fthere is not a Default dial string set against the phone then the user will receive dial tone when the line is manually selected Al gt Page 39 F3 gt Page 64 Initial Digit Timer This timer determines the delay after which the phone goes back
140. y the phone will be kept up to date about the current survivability state even after a restart Survivability with multiple geographically separated SIP Servers Another way to realize survivability is the use of multiple geographically separated SIP servers Normally the phone is registered with that server that has the highest priority in the DNS SRV server list If the highest priority server fails to respond the phone will register with the server that has the second highest priority Use of a Backup SIP Server Along with the registration at the primary SIP server the phone is registered with a backup SIP server In normal operation the phone uses the primary server for outgoing calls If the phone de tects that the connection to the primary SIP server is lost it uses the backup server for outgoing calls Parameter Web Interface path Menu Page 60 Backup Address W gt Page 119 Survivability i Backup IP address or DNS name enter IP address Backup Port W gt Page 119 Survivability z Port enter port address Backup Registration E W gt Page 119 Survivability Backup Registration mark to enable Backup Reg Timer E W gt Page 119 SIP environment x Backup Registration timer value enter time Backup OBP W gt Page 119 Survivability Backup Outbound proxy mark to enable Backup Transport e W gt Page 119 Survivability Backup SIP transport TCP or UDP UDP
141. y allowed Mark Payload security allowed as on to make sure you set up a secure connection whenever possible A gt Page 52 3 gt Page 77 Ping Run this gt PING test to check whether a server or another terminal device e g the OpenStage 5 SIP or servers can be reached by gt IP or domain name For this enter or select an gt IP address or domain name as a test target the connection to which you wish to test Value range table for user specified IP Permitted values numeric with DNS also alphanumeric Length max 15 digits incl dots with DNS also 92 digits Al gt Page 51 53 gt Page 74 Play DTMF RFC2833 Playback DTMF tones when received as RTP event according to RFC2833 Playback of DTMF tones is disabled by default DTMF tones will be played when you enter a conference assumed the function is enabled espe cially in conference server systems Al gt Page 50 3 gt Page 66 Port Control This parameters are needed for development only The following options are available Service Agent e Testlnterface SNMP Port Al Page 54 1 gt Page 79 Primary DNS IP address Enter the gt IP address of the gt DNS server if not provided by gt DHCP dynamically gt DHCP IP assignment Value range table Permitted values numeric Length max 15 digits incl dots Al gt Page 35 F3 gt Page 62 QDC Address Enter IP address of the Quality Data Collection Server F
142. yment service DLS for configuring phones Web interface for configuring individual phones SNMP Speech Support for G711 U and A Law G723 and G729 High Quality speaker phone functionality G711 Silence Suppression Audio codec G 722 offers a wider audio bandwidth resulting in major improvement in the repre sented speech quality 17 Phone Features Call Features Call forwarding Unconditional On Busy On no Reply Call waiting Consultation Unattended Transfer Attended Transfer Join Do not Disturb Hold Message Waiting MultiLine Call back More features available with server related access code DTMF The phone provides 2 mechanisms for transmitting 2 DTMF information inband and DTMF in RTP see RFC 2833 The phone does not support outband DTMF through SIP messaging There are no configuration parameters on the phone which control the use of DTMF A process of negotiation is used during call setup to determine which form of DTMF signaling will be used The phone supports send DTMF information in response to the user pressing the keys 0 9 and and when in a call connected state When a call is made from a phone it will Offer the remote endpoint support for DTMF in RTP this is carried in the SDP protocol If the far end does not answer that it can support DTMF in RTP then DTMF in band will be used otherwise DTMF in RTP will be used When DTMF in RTP is negotiated the phone will always

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